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Subject: [Sip] I-D Action:draft-ietf-sip-media-security-requirements-07.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.


	Title           : Requirements and Analysis of Media Security Management Protocols
	Author(s)       : D. Wing, et al.
	Filename        : draft-ietf-sip-media-security-requirements-07.txt
	Pages           : 47
	Date            : 2008-06-02

This document describes requirements for a protocol to negotiate a
security context for SIP-signaled SRTP media.  In addition to the
natural security requirements, this negotiation protocol must
interoperate well with SIP in certain ways.  A number of proposals
have been published and a summary of these proposals is in the
appendix of this document.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-media-security-requirements-07.txt

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_______________________________________________
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I just posted draft-ietf-sip-media-security-requirements-07.txt which
incorporates the text suggested by Eric in
<http://www.ietf.org/mail-archive/web/sip/current/msg23495.html> and Steffen's
change to Eric's text described in
<http://www.ietf.org/mail-archive/web/sip/current/msg23502.html>.

With these changes, we now have:

   ...
   4.9.  Certificates

   The discussion in this section relates to R-CERTS.

   On the Internet and on some private networks, validating another
   peer's certificate is often done through a trust anchor -- a list of
   Certificate Authorities that are trusted.  It can be difficult or
   expensive for a peer to obtain these certificates.  In all cases,
   both parties to the call would need to trust the same trust anchor
   (i.e., "certificate authority").  For these reasons, it is important
   that the media plane key management protocol offer a mechanism that
   allows end-users who have no prior association to authenticate to
   each other without acquiring credentials from a third party trust
   point.  Note that this does not rule out mechanisms in which servers
   have certificates and attest to the identities of end-users.

   ...

   5.2.  Security Requirements
   ...
   R-CERTS:
         The key management protocol MUST NOT require that end-users
         obtain credentials (certificates or private keys) from a third-
         party trust anchor.
   ...

Side-by-side diffs:
  http://tinyurl.com/3kc85k


This closes the open issues with this document and it is now ready to forward
to the IESG.
-d

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


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Hi Guys

   I have a question as following

A and B are already connented , A sent a re-invite to B, and the three-way
handshake  is ok
after it, the rtp (A to B)'s seq no is larger than the before  and the SSRC
is different I think it is ok , but the   timestamp is samller than
the before I do not know whther it is ok or not

can anyone give me some advices

Thanks

Samman

2008-6-3

-- 
msn:yongguangsun@hotmail.com <msn%3Ayongguangsun@hotmail.com>

------=_Part_533_14713905.1212480782684
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<div>Hi Guys</div>
<div>&nbsp;</div>
<div>&nbsp;&nbsp; I have a question as following</div>
<div>&nbsp;</div>
<div>A and B are already&nbsp;connented , A sent a re-invite to B,&nbsp;and the three-way handshake &nbsp;is ok </div>
<div>after it, the&nbsp;rtp&nbsp;(A to B)&#39;s&nbsp;seq no is larger than the before&nbsp; and the SSRC is different I think it is ok&nbsp;, but the &nbsp;&nbsp;timestamp is samller than </div>
<div>the before I do not know whther it is ok or not </div>
<div>&nbsp;</div>
<div>can anyone give me some advices</div>
<div>&nbsp;</div>
<div>Thanks</div>
<div>&nbsp;</div>
<div>Samman</div>
<div>&nbsp;</div>
<div>2008-6-3<br clear="all"><br>-- <br><a href="mailto:msn%3Ayongguangsun@hotmail.com">msn:yongguangsun@hotmail.com</a> </div>

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_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============1546956562==--


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From: "CHEN JIANBIN" <Jianbin.Chen@alcatel-sbell.com.cn>
To: =?gb2312?B?y+/TwLni?= <sammanengineer@gmail.com>, <sip@ietf.org>,
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Subject: Re: [Sip] rtp seq and timestamp
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Samman,
=20
According to RFC3550, SSRC means Synchronization Source.
     "All packets from a synchronization source form part of the same
      timing and sequence number space, so a receiver groups packets by
      synchronization source for playback. "
=20
For the SSRC is changed as you mentioned, the "seq no" and "timestamp" =
has no relationship with the previous one.
Different SSRC means different timing and sequence number space.
It has no meaning to compare seq no and timestamp with different SSRC.=20
=20
BR
Ben
________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =
=CB=EF=D3=C0=B9=E2
Sent: 2008=C4=EA6=D4=C23=C8=D5 16:13
To: sip@ietf.org; IETF Sipping List
Subject: [Sip] rtp seq and timestamp


Hi Guys
=20
   I have a question as following
=20
A and B are already connented , A sent a re-invite to B, and the =
three-way handshake  is ok=20
after it, the rtp (A to B)'s seq no is larger than the before  and the =
SSRC is different I think it is ok , but the   timestamp is samller than =

the before I do not know whther it is ok or not=20
=20
can anyone give me some advices
=20
Thanks
=20
Samman
=20
2008-6-3

--=20
msn:yongguangsun@hotmail.com <mailto:msn%3Ayongguangsun@hotmail.com> =20

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<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; charset=3Dgb2312">
<META content=3D"MSHTML 6.00.2900.3314" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D966513006-04062008><FONT =
face=3D"Trebuchet MS"=20
color=3D#0000ff>Samman,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D966513006-04062008><FONT =
face=3D"Trebuchet MS"=20
color=3D#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D966513006-04062008><FONT =
face=3D"Trebuchet MS"=20
color=3D#0000ff>According to RFC3550,&nbsp;SSRC&nbsp;means =
Synchronization=20
Source.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D966513006-04062008><FONT =
face=3D"Trebuchet MS"=20
color=3D#0000ff>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;"All packets from a =
synchronization=20
source form part of the same<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; timing =
and=20
sequence number space, so a receiver groups packets=20
by<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; synchronization source for=20
playback.&nbsp;"</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D966513006-04062008><FONT =
face=3D"Trebuchet MS"=20
color=3D#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D966513006-04062008><FONT =
face=3D"Trebuchet MS"=20
color=3D#0000ff>For the SSRC is changed as you mentioned, the "seq no" =
and=20
"timestamp"&nbsp;has no relationship with the previous =
one.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D966513006-04062008><FONT =
face=3D"Trebuchet MS"=20
color=3D#0000ff>Different SSRC means different timing and sequence =
number=20
space.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D966513006-04062008><FONT =
face=3D"Trebuchet MS"=20
color=3D#0000ff>It&nbsp;has no meaning to compare&nbsp;seq no and =
timestamp with=20
different SSRC.</FONT>&nbsp;</SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D966513006-04062008><FONT =
face=3D"Trebuchet MS"=20
color=3D#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D966513006-04062008><FONT =
face=3D"Trebuchet MS"=20
color=3D#0000ff>BR</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D966513006-04062008><FONT =
face=3D"Trebuchet MS"=20
color=3D#0000ff>Ben</FONT></SPAN></DIV>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
[mailto:sip-bounces@ietf.org] <B>On Behalf Of =
</B>=CB=EF=D3=C0=B9=E2<BR><B>Sent:</B> 2008=C4=EA6=D4=C23=C8=D5=20
16:13<BR><B>To:</B> sip@ietf.org; IETF Sipping List<BR><B>Subject:</B> =
[Sip] rtp=20
seq and timestamp<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV>Hi Guys</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;&nbsp; I have a question as following</DIV>
<DIV>&nbsp;</DIV>
<DIV>A and B are already&nbsp;connented , A sent a re-invite to =
B,&nbsp;and the=20
three-way handshake &nbsp;is ok </DIV>
<DIV>after it, the&nbsp;rtp&nbsp;(A to B)'s&nbsp;seq no is larger than =
the=20
before&nbsp; and the SSRC is different I think it is ok&nbsp;, but the=20
&nbsp;&nbsp;timestamp is samller than </DIV>
<DIV>the before I do not know whther it is ok or not </DIV>
<DIV>&nbsp;</DIV>
<DIV>can anyone give me some advices</DIV>
<DIV>&nbsp;</DIV>
<DIV>Thanks</DIV>
<DIV>&nbsp;</DIV>
<DIV>Samman</DIV>
<DIV>&nbsp;</DIV>
<DIV>2008-6-3<BR clear=3Dall><BR>-- <BR><A=20
href=3D"mailto:msn%3Ayongguangsun@hotmail.com">msn:yongguangsun@hotmail.c=
om</A>=20
</DIV></BODY></HTML>

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Use sipping@ietf.org for new developments on the application of sip
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From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
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(As SIP WG cochair)

This has been raised in the SIP group a couple of times, and we have not
yet gained an idea of whether to proceed with it in the SIP WG or not.
 
There have been a couple of technical comments raised in the past from
the security experts; my understanding is that these have now been
clarified.
 
I believe in the past there has been an interested community that says
this is useful. It has also been clarified that 3GPP would like to use
it in their specifications.
 
So could I ask the WG to look at this document, and formally to indicate
whether they believe some work with this scope should be progressed as a
SIP WG item.
 
http://www.ietf.org/internet-drafts/draft-dotson-sip-mutual-auth-02.txt

So if you support it please indicate to the SIP WG chairs.

If you have technical concerns (i.e. it should not proceed because it is
technically flawed beyond fixing) then please ideally mail the list,
although the SIP WG chairs will also accept input.

Responses please in 1 calendar week from this date.

Regards

Keith


________________________________

	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
Behalf Of Stuart Hoggan
	Sent: Friday, May 02, 2008 5:15 PM
	To: sip@ietf.org
	Subject: [Sip] draft-dotson-sip-mutual-auth-02
	
	

	Folks,

	 

	Based on the feedback received in Philadelphia, we have updated
draft-dotson-sip-mutual-auth. The changes clarify the usage of the
Proxy-Authentication-Info header, including support for multiple values.


	 

	Any further comments or suggestions will be appreciated. It
would also be nice to obtain WG feedback on pursuing this I-D as a WG
document.

	 

	Regards,

	Stuart

	 

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Subject: [Sip] What is the port number in "Invite" request-line? Thanks
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Dear all

=20

INVITE sip:bob@biloxi.com SIP/2.0:5555

INVITE sip:bob@biloxi.com SIP/2.0

=20

Above are 2 "Invite" request-line examples. In example one, it indicates
a port number, 5555. What does the port number mean?=20

=20

Is it the port number of SIP proxy? Why we include port number here? The
SIP proxy(biloxi.com) application layer who receives this "Invite"
should receive this via port 5555 from lowe layer(UDP). Therefore, The
SIP proxy(biloxi.com) application layer already know the port is 5555.
Actualy, the SIP proxy application layer keeps listening port 5555.=20

=20

The second "Invite" request-line example is from RFC3261 and does not
include port number. Can I say the port number is useless? So why
example one includes a useless part in "Invite" request-line URL?=20

=20

Or is the port number is which the UAS is listening to? How the UAC
knows it is 5555 if so?=20

=20

Any standard on this port number issue? Thanks.=20

=20

Best regards

Steven


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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'>Dear =
all<o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>INVITE sip:bob@biloxi.com =
SIP/2.0:5555<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'>INVITE sip:bob@biloxi.com =
SIP/2.0<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'>Above are 2 &#8220;Invite&#8221; =
request-line
examples. In example one, it indicates a port number, 5555. What does =
the port number
mean? <o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Is it the port number of SIP proxy? =
Why we
include port number here? The SIP proxy(biloxi.com) application layer =
who
receives this &quot;Invite&quot; should receive this via port 5555 from =
lowe
layer(UDP). Therefore, The SIP proxy(biloxi.com) application layer =
already know
the port is 5555. Actualy, the SIP proxy application layer keeps =
listening port
5555. <o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>The second &#8220;Invite&#8221; =
request-line
example is from RFC3261 and does not include port number. Can I say the =
port
number is useless? So why example one includes a useless part in
&#8220;Invite&#8221; request-line URL? <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 is the port number is which the =
UAS is
listening to? How the UAC knows it is 5555 if so? =
<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'>Any standard on this port number =
issue? Thanks.
<o:p></o:p></span></font></p>

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

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

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Thread-Topic: [Sip] What is the port number in "Invite" request-line? Thanks
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From: "Attila Sipos" <attila.sipos@vegastream.com>
To: "Yuantao Zhang" <yuantao.zhang@ericsson.com>, <sip@ietf.org>,
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Subject: Re: [Sip] What is the port number in "Invite" request-line? Thanks
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=20
It is possible to have two instances of a UA (for example) on one unit.

So using the port is theoretically one way to differentiate between
instances.
=20
It's not useless.  If you get a Contact header with a port then you have
to send to that port (discounting NAT traversal scenarios) otherwise
everything will break.
=20
>>Or is the port number is which the UAS is listening to? How the UAC
knows it is 5555 if so?=20
=20
A UAC might not know but a proxy/registrar might - for example if a user
was registered with
the Contact as port 5555, then requests to that user will get forwarded
to port 5555.
Another user using, for example, port 5556 will register using a Contact
with 5556
and so will receive requests to that port.
=20
Regards,

Attila
=20
=20
________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Yuantao Zhang
Sent: 04 June 2008 11:10
To: sip@ietf.org; discussion@sipforum.org
Subject: [Sip] What is the port number in "Invite" request-line? Thanks



Dear all

INVITE sip:bob@biloxi.com SIP/2.0:5555

INVITE sip:bob@biloxi.com SIP/2.0

Above are 2 "Invite" request-line examples. In example one, it indicates
a port number, 5555. What does the port number mean?=20

Is it the port number of SIP proxy? Why we include port number here? The
SIP proxy(biloxi.com) application layer who receives this "Invite"
should receive this via port 5555 from lowe layer(UDP). Therefore, The
SIP proxy(biloxi.com) application layer already know the port is 5555.
Actualy, the SIP proxy application layer keeps listening port 5555.=20

The second "Invite" request-line example is from RFC3261 and does not
include port number. Can I say the port number is useless? So why
example one includes a useless part in "Invite" request-line URL?=20

Or is the port number is which the UAS is listening to? How the UAC
knows it is 5555 if so?=20

Any standard on this port number issue? Thanks.=20

Best regards

Steven


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defanghtml_xmlns:o=3D"urn:schemas-microsoft-com:office:office"=20
defanghtml_xmlns:v=3D"urn:schemas-microsoft-com:vml"><HEAD><TITLE>MUST 200 =
OK contain SDP? Thanks</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.2873" name=3DGENERATOR>
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<BODY lang=3DEN-US vLink=3Dred link=3Dblue>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D390501410-04062008><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>It is possible to have two instances of a UA (for =
example)=20
</FONT></SPAN><SPAN class=3D390501410-04062008><FONT face=3DArial color=3D#=
0000ff=20
size=3D2>on one unit.</FONT></SPAN></DIV><SPAN class=3D390501410-04062008>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2></F=
ONT><FONT=20
face=3DArial color=3D#0000ff size=3D2></FONT><FONT face=3DArial color=3D#00=
00ff=20
size=3D2></FONT><FONT face=3DArial color=3D#0000ff size=3D2></FONT><BR><FON=
T=20
face=3DArial><FONT color=3D#0000ff><FONT size=3D2>So using the port is&nbsp=
;<SPAN=20
class=3D390501410-04062008>theoretically </SPAN>one way to=20
differentiate&nbsp;<SPAN class=3D390501410-04062008>between=20
instances.</SPAN></FONT></FONT></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT>&nbsp;</DIV></SPAN>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D390501410-04062008><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>It's not useless.&nbsp; If you get a Contact heade=
r=20
with&nbsp;a port then&nbsp;you have</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D390501410-04062008><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D390501410-04062008><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>to send to that port (discounting NAT traversal sc=
enarios)=20
otherwise everything will break.</FONT></SPAN></DIV></FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial><FONT color=3D#000080><FONT =
size=3D2><SPAN=20
class=3D390501410-04062008></SPAN></FONT></FONT></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial><FONT color=3D#000080><FONT =
size=3D2><SPAN=20
class=3D390501410-04062008>&gt;&gt;</SPAN>Or is the port number is which th=
e UAS=20
is listening to? How the UAC knows it is 5555 if so? </FONT></FONT></FONT><=
/DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><SPAN class=3D390501410-04062008>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D390501410-04062008><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>A UAC might not know but a proxy/registrar might -=
 for=20
example if a user was registered with</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D390501410-04062008><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>the Contact as port 5555, then requests to that us=
er will=20
get forwarded to port 5555.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D390501410-04062008><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Another user using, for example, port&nbsp;5556 wi=
ll=20
register using a Contact with 5556</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D390501410-04062008><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>and so will receive requests to that=20
port.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D390501410-04062008><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial><FONT size=3D2><FONT color=
=3D#0000ff>R<SPAN=20
class=3D390501410-04062008>egards,</SPAN></FONT></FONT></FONT></DIV><FONT><=
SPAN=20
class=3D390501410-04062008>
<DIV dir=3Dltr align=3Dleft><BR><FONT face=3DArial><FONT color=3D#0000ff><F=
ONT=20
size=3D2>Attil<SPAN class=3D390501410-04062008>a</SPAN></FONT></FONT></FONT=
></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial><FONT color=3D#0000ff><FONT =
size=3D2><SPAN=20
class=3D390501410-04062008></SPAN></FONT></FONT></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft></SPAN></FONT></SPAN>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
[mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Yuantao Zhang<BR><B>Sent:=
</B>=20
04 June 2008 11:10<BR><B>To:</B> sip@ietf.org;=20
discussion@sipforum.org<BR><B>Subject:</B> [Sip] What is the port number in=
=20
"Invite" request-line? Thanks<BR></FONT><BR></DIV></DIV>
<DIV></DIV>
<DIV class=3DSection1>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><DEFANGHTML_S=
PAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Dear=20
all<O:P></O:P></DEFANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><DEFANGHTML_S=
PAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><O:P></O:P></DEF=
ANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><DEFANGHTML_S=
PAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">INVITE=20
sip:bob@biloxi.com SIP/2.0:5555<O:P></O:P></DEFANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><DEFANGHTML_S=
PAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">INVITE=20
sip:bob@biloxi.com SIP/2.0<O:P></O:P></DEFANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><DEFANGHTML_S=
PAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><O:P></O:P></DEF=
ANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><DEFANGHTML_S=
PAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Above are 2 &#82=
20;Invite&#8221;=20
request-line examples. In example one, it indicates a port number, 5555. Wh=
at=20
does the port number mean? <O:P></O:P></DEFANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><DEFANGHTML_S=
PAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><O:P></O:P></DEF=
ANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><DEFANGHTML_S=
PAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Is it the port n=
umber=20
of SIP proxy? Why we include port number here? The SIP proxy(biloxi.com)=20
application layer who receives this "Invite" should receive this via port 5=
555=20
from lowe layer(UDP). Therefore, The SIP proxy(biloxi.com) application laye=
r=20
already know the port is 5555. Actualy, the SIP proxy application layer kee=
ps=20
listening port 5555. <O:P></O:P></DEFANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><DEFANGHTML_S=
PAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><O:P></O:P></DEF=
ANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><DEFANGHTML_S=
PAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">The second &#822=
0;Invite&#8221;=20
request-line example is from RFC3261 and does not include port number. Can =
I say=20
the port number is useless? So why example one includes a useless part in=
=20
&#8220;Invite&#8221; request-line URL? <O:P></O:P></DEFANGHTML_SPAN></FONT>=
</P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><DEFANGHTML_S=
PAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><O:P></O:P></DEF=
ANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><DEFANGHTML_S=
PAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Or is the port n=
umber=20
is which the UAS is listening to? How the UAC knows it is 5555 if so?=20
<O:P></O:P></DEFANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><DEFANGHTML_S=
PAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><O:P></O:P></DEF=
ANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><DEFANGHTML_S=
PAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Any standard on =
this=20
port number issue? Thanks. <O:P></O:P></DEFANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><DEFANGHTML_S=
PAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><O:P></O:P></DEF=
ANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><DEFANGHTML_S=
PAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Best=20
regards<O:P></O:P></DEFANGHTML_SPAN></FONT></P>
<P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><DEFANGHTML_S=
PAN=20
style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Steven<O:P></O:P=
></DEFANGHTML_SPAN></FONT></P></DIV><br><br>
<P align=3Dleft><FONT style=3D"BACKGROUND-COLOR: #ffffff">.</FONT></P>
</body></HTML>

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Cc: sip@ietf.org, discussion@sipforum.org,
	Yuantao Zhang <yuantao.zhang@ericsson.com>
Subject: Re: [Sip] What is the port number in "Invite" request-line? Thanks
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Attila Sipos wrote:
>  
> It is possible to have two instances of a UA (for example) on one unit.
> 
> So using the port is theoretically one way to differentiate between 
> instances.
>  
> It's not useless.  If you get a Contact header with a port then you have
> to send to that port (discounting NAT traversal scenarios) otherwise 
> everything will break.
>  
>> >Or is the port number is which the UAS is listening to? How the UAC 
> knows it is 5555 if so?
>  
> A UAC might not know but a proxy/registrar might - for example if a user 
> was registered with
> the Contact as port 5555, then requests to that user will get forwarded 
> to port 5555.
> Another user using, for example, port 5556 will register using a Contact 
> with 5556
> and so will receive requests to that port.

Right. This illustrates a good point:

In general a UAC should not be expected to know "what port a UAS listens 
on". All A UAC should be doing is sending requests to URIs that it has 
been given. If the URI it is given contains a port number, then it 
should honor that.

This gets bent out of shape a bit by numeric dialing, where the UAC is 
only given a dial string and must derive a URI from it. The "standard" 
way to do this is via ENUM, where effectively the numeric address is 
looked up in a DB to obtain a URI. In that case again the UAC has been 
"given" the URI, which may have a port or not.

In the non-standard but common case where the UAC *constructs* a URI 
from a dial string, the UAC must still be *told* what to use for the 
remainder of the URI.

	Paul
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From sip-bounces@ietf.org  Wed Jun  4 11:08:14 2008
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From: "Francois Audet" <audet@nortel.com>
To: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>, <sip@ietf.org>
Subject: Re: [Sip] draft-dotson-sip-mutual-auth-02
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I do not believe that this document provide enought 
justification of what is the problem that is being attempted to
be solved, and why it isn't already solved by existing mechanism
that are widely implemented (such as TLS).

Section 9 (Security Considerations) I believe touches on it
a little bit, but it seems both incomplete and out of place.

The problem statement needs to be brough forward in the document
(say to section 3 or a new section). The scope of applicability
of the solution should also be described up-front).

I wan't to clarfiy I'm not necessarily against this: I just want
the justification to be clearer. The onus should be on this
document to clearly demonstrate why what we currently have
is broken.

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of DRAGE, Keith (Keith)
> Sent: Wednesday, June 04, 2008 02:48
> To: sip@ietf.org
> Subject: Re: [Sip] draft-dotson-sip-mutual-auth-02
> 
> (As SIP WG cochair)
> 
> This has been raised in the SIP group a couple of times, and 
> we have not yet gained an idea of whether to proceed with it 
> in the SIP WG or not.
>  
> There have been a couple of technical comments raised in the 
> past from the security experts; my understanding is that 
> these have now been clarified.
>  
> I believe in the past there has been an interested community 
> that says this is useful. It has also been clarified that 
> 3GPP would like to use it in their specifications.
>  
> So could I ask the WG to look at this document, and formally 
> to indicate whether they believe some work with this scope 
> should be progressed as a SIP WG item.
>  
> http://www.ietf.org/internet-drafts/draft-dotson-sip-mutual-au
> th-02.txt
> 
> So if you support it please indicate to the SIP WG chairs.
> 
> If you have technical concerns (i.e. it should not proceed 
> because it is technically flawed beyond fixing) then please 
> ideally mail the list, although the SIP WG chairs will also 
> accept input.
> 
> Responses please in 1 calendar week from this date.
> 
> Regards
> 
> Keith
> 
> 
> ________________________________
> 
> 	From: sip-bounces@ietf.org 
> [mailto:sip-bounces@ietf.org] On Behalf Of Stuart Hoggan
> 	Sent: Friday, May 02, 2008 5:15 PM
> 	To: sip@ietf.org
> 	Subject: [Sip] draft-dotson-sip-mutual-auth-02
> 	
> 	
> 
> 	Folks,
> 
> 	 
> 
> 	Based on the feedback received in Philadelphia, we have 
> updated draft-dotson-sip-mutual-auth. The changes clarify the 
> usage of the Proxy-Authentication-Info header, including 
> support for multiple values.
> 
> 
> 	 
> 
> 	Any further comments or suggestions will be 
> appreciated. It would also be nice to obtain WG feedback on 
> pursuing this I-D as a WG document.
> 
> 	 
> 
> 	Regards,
> 
> 	Stuart
> 
> 	 
> 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use 
> sip-implementors@cs.columbia.edu for questions on current sip 
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
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From sip-bounces@ietf.org  Wed Jun  4 11:33:31 2008
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From: "Sumanth Channabasappa" <sumanth@cablelabs.com>
To: "Francois Audet" <audet@nortel.com>,
	"DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>, <sip@ietf.org>
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Francois,

Thanks for your thoughts. We can certainly add more details if it is
unclear. 

As a clarification, I don't think we are saying that the current
mechanism is necessarily broken (which is why it is not an essential
correction, as discussed earlier in the WG). However, we want to allow
for the use of this header in deployments that use Digest (with or
without TLS; e.g., when integrity protection is provided by other means)
and can benefit from its advantages (mutual auth, nextnonce). 

- S

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Francois Audet
Sent: Wednesday, June 04, 2008 12:08 PM
To: DRAGE, Keith (Keith); sip@ietf.org
Subject: Re: [Sip] draft-dotson-sip-mutual-auth-02

I do not believe that this document provide enought 
justification of what is the problem that is being attempted to
be solved, and why it isn't already solved by existing mechanism
that are widely implemented (such as TLS).

Section 9 (Security Considerations) I believe touches on it
a little bit, but it seems both incomplete and out of place.

The problem statement needs to be brough forward in the document
(say to section 3 or a new section). The scope of applicability
of the solution should also be described up-front).

I wan't to clarfiy I'm not necessarily against this: I just want
the justification to be clearer. The onus should be on this
document to clearly demonstrate why what we currently have
is broken.

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of DRAGE, Keith (Keith)
> Sent: Wednesday, June 04, 2008 02:48
> To: sip@ietf.org
> Subject: Re: [Sip] draft-dotson-sip-mutual-auth-02
> 
> (As SIP WG cochair)
> 
> This has been raised in the SIP group a couple of times, and 
> we have not yet gained an idea of whether to proceed with it 
> in the SIP WG or not.
>  
> There have been a couple of technical comments raised in the 
> past from the security experts; my understanding is that 
> these have now been clarified.
>  
> I believe in the past there has been an interested community 
> that says this is useful. It has also been clarified that 
> 3GPP would like to use it in their specifications.
>  
> So could I ask the WG to look at this document, and formally 
> to indicate whether they believe some work with this scope 
> should be progressed as a SIP WG item.
>  
> http://www.ietf.org/internet-drafts/draft-dotson-sip-mutual-au
> th-02.txt
> 
> So if you support it please indicate to the SIP WG chairs.
> 
> If you have technical concerns (i.e. it should not proceed 
> because it is technically flawed beyond fixing) then please 
> ideally mail the list, although the SIP WG chairs will also 
> accept input.
> 
> Responses please in 1 calendar week from this date.
> 
> Regards
> 
> Keith
> 
> 
> ________________________________
> 
> 	From: sip-bounces@ietf.org 
> [mailto:sip-bounces@ietf.org] On Behalf Of Stuart Hoggan
> 	Sent: Friday, May 02, 2008 5:15 PM
> 	To: sip@ietf.org
> 	Subject: [Sip] draft-dotson-sip-mutual-auth-02
> 	
> 	
> 
> 	Folks,
> 
> 	 
> 
> 	Based on the feedback received in Philadelphia, we have 
> updated draft-dotson-sip-mutual-auth. The changes clarify the 
> usage of the Proxy-Authentication-Info header, including 
> support for multiple values.
> 
> 
> 	 
> 
> 	Any further comments or suggestions will be 
> appreciated. It would also be nice to obtain WG feedback on 
> pursuing this I-D as a WG document.
> 
> 	 
> 
> 	Regards,
> 
> 	Stuart
> 
> 	 
> 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use 
> sip-implementors@cs.columbia.edu for questions on current sip 
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


From sip-bounces@ietf.org  Wed Jun  4 11:36:59 2008
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From: Dean Willis <dean.willis@softarmor.com>
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Subject: [Sip] Conclusion of WGLC for Outbound: draft-ietf-sip-outbound-14
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The latest Outbound draft:

http://www.ietf.org/internet-drafts/draft-ietf-sip-outbound-14.txt

supposedly accounts for all issues raised in the rather lengthy (it  
started July 9, 2007) working group last call of the draft.

I expect to be writing this up and sending it to the IESG shortly (as  
in the end of this week), so if you have any last comments, get them  
in now.

--
Dean (as chair and assigned shepherd of this wayward sheep)
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From sip-bounces@ietf.org  Wed Jun  4 12:04:17 2008
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Thread-Topic: [Sip] draft-dotson-sip-mutual-auth-02
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From: "Francois Audet" <audet@nortel.com>
To: "Sumanth Channabasappa" <sumanth@cablelabs.com>,
	"DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>, <sip@ietf.org>
Subject: Re: [Sip] draft-dotson-sip-mutual-auth-02
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Ok, thanks.

I would like the begining of the document (say section 3, or a Scope,
Applicability statement or similar section) to explains
when and why this is useful.

The purpose is really for an implementor to be able by reading this 
section to be in a position to determine if it's somthing that should
he needs to worry about or not. 

So, say if I'm developping a UA for an IMS environment versus an
Enteprise SIP UA, or an Enteprise SIP access to a service provider
network, I'll know if this applies to me or not.

Cheers. 

> -----Original Message-----
> From: Sumanth Channabasappa [mailto:sumanth@cablelabs.com] 
> Sent: Wednesday, June 04, 2008 11:33
> To: Audet, Francois (SC100:3055); DRAGE, Keith (Keith); sip@ietf.org
> Subject: RE: [Sip] draft-dotson-sip-mutual-auth-02
> 
> Francois,
> 
> Thanks for your thoughts. We can certainly add more details 
> if it is unclear. 
> 
> As a clarification, I don't think we are saying that the 
> current mechanism is necessarily broken (which is why it is 
> not an essential correction, as discussed earlier in the WG). 
> However, we want to allow for the use of this header in 
> deployments that use Digest (with or without TLS; e.g., when 
> integrity protection is provided by other means) and can 
> benefit from its advantages (mutual auth, nextnonce). 
> 
> - S
> 
> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of Francois Audet
> Sent: Wednesday, June 04, 2008 12:08 PM
> To: DRAGE, Keith (Keith); sip@ietf.org
> Subject: Re: [Sip] draft-dotson-sip-mutual-auth-02
> 
> I do not believe that this document provide enought 
> justification of what is the problem that is being attempted 
> to be solved, and why it isn't already solved by existing 
> mechanism that are widely implemented (such as TLS).
> 
> Section 9 (Security Considerations) I believe touches on it a 
> little bit, but it seems both incomplete and out of place.
> 
> The problem statement needs to be brough forward in the 
> document (say to section 3 or a new section). The scope of 
> applicability of the solution should also be described up-front).
> 
> I wan't to clarfiy I'm not necessarily against this: I just 
> want the justification to be clearer. The onus should be on 
> this document to clearly demonstrate why what we currently 
> have is broken.
> 
> > -----Original Message-----
> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of 
> > DRAGE, Keith (Keith)
> > Sent: Wednesday, June 04, 2008 02:48
> > To: sip@ietf.org
> > Subject: Re: [Sip] draft-dotson-sip-mutual-auth-02
> > 
> > (As SIP WG cochair)
> > 
> > This has been raised in the SIP group a couple of times, 
> and we have 
> > not yet gained an idea of whether to proceed with it in the 
> SIP WG or 
> > not.
> >  
> > There have been a couple of technical comments raised in 
> the past from 
> > the security experts; my understanding is that these have now been 
> > clarified.
> >  
> > I believe in the past there has been an interested 
> community that says 
> > this is useful. It has also been clarified that 3GPP would 
> like to use 
> > it in their specifications.
> >  
> > So could I ask the WG to look at this document, and formally to 
> > indicate whether they believe some work with this scope should be 
> > progressed as a SIP WG item.
> >  
> > http://www.ietf.org/internet-drafts/draft-dotson-sip-mutual-au
> > th-02.txt
> > 
> > So if you support it please indicate to the SIP WG chairs.
> > 
> > If you have technical concerns (i.e. it should not proceed 
> because it 
> > is technically flawed beyond fixing) then please ideally mail the 
> > list, although the SIP WG chairs will also accept input.
> > 
> > Responses please in 1 calendar week from this date.
> > 
> > Regards
> > 
> > Keith
> > 
> > 
> > ________________________________
> > 
> > 	From: sip-bounces@ietf.org
> > [mailto:sip-bounces@ietf.org] On Behalf Of Stuart Hoggan
> > 	Sent: Friday, May 02, 2008 5:15 PM
> > 	To: sip@ietf.org
> > 	Subject: [Sip] draft-dotson-sip-mutual-auth-02
> > 	
> > 	
> > 
> > 	Folks,
> > 
> > 	 
> > 
> > 	Based on the feedback received in Philadelphia, we have updated 
> > draft-dotson-sip-mutual-auth. The changes clarify the usage of the 
> > Proxy-Authentication-Info header, including support for multiple 
> > values.
> > 
> > 
> > 	 
> > 
> > 	Any further comments or suggestions will be 
> appreciated. It would 
> > also be nice to obtain WG feedback on pursuing this I-D as a WG 
> > document.
> > 
> > 	 
> > 
> > 	Regards,
> > 
> > 	Stuart
> > 
> > 	 
> > 
> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use 
> > sip-implementors@cs.columbia.edu for questions on current sip Use 
> > sipping@ietf.org for new developments on the application of sip
> > 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use 
> sip-implementors@cs.columbia.edu for questions on current sip 
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


From sip-bounces@ietf.org  Wed Jun  4 12:42:50 2008
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From: "Sumanth Channabasappa" <sumanth@cablelabs.com>
To: "Francois Audet" <audet@nortel.com>,
	"DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>, <sip@ietf.org>
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Subject: Re: [Sip] draft-dotson-sip-mutual-auth-02
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Sounds fair, we can provide additional information in a revision
(probably when the WG decides to pursue this effort?).

Thanks!
- S

-----Original Message-----
From: Francois Audet [mailto:audet@nortel.com] 
Sent: Wednesday, June 04, 2008 1:02 PM
To: Sumanth Channabasappa; DRAGE, Keith (Keith); sip@ietf.org
Subject: RE: [Sip] draft-dotson-sip-mutual-auth-02

Ok, thanks.

I would like the begining of the document (say section 3, or a Scope,
Applicability statement or similar section) to explains
when and why this is useful.

The purpose is really for an implementor to be able by reading this 
section to be in a position to determine if it's somthing that should
he needs to worry about or not. 

So, say if I'm developping a UA for an IMS environment versus an
Enteprise SIP UA, or an Enteprise SIP access to a service provider
network, I'll know if this applies to me or not.

Cheers. 

> -----Original Message-----
> From: Sumanth Channabasappa [mailto:sumanth@cablelabs.com] 
> Sent: Wednesday, June 04, 2008 11:33
> To: Audet, Francois (SC100:3055); DRAGE, Keith (Keith); sip@ietf.org
> Subject: RE: [Sip] draft-dotson-sip-mutual-auth-02
> 
> Francois,
> 
> Thanks for your thoughts. We can certainly add more details 
> if it is unclear. 
> 
> As a clarification, I don't think we are saying that the 
> current mechanism is necessarily broken (which is why it is 
> not an essential correction, as discussed earlier in the WG). 
> However, we want to allow for the use of this header in 
> deployments that use Digest (with or without TLS; e.g., when 
> integrity protection is provided by other means) and can 
> benefit from its advantages (mutual auth, nextnonce). 
> 
> - S
> 
> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of Francois Audet
> Sent: Wednesday, June 04, 2008 12:08 PM
> To: DRAGE, Keith (Keith); sip@ietf.org
> Subject: Re: [Sip] draft-dotson-sip-mutual-auth-02
> 
> I do not believe that this document provide enought 
> justification of what is the problem that is being attempted 
> to be solved, and why it isn't already solved by existing 
> mechanism that are widely implemented (such as TLS).
> 
> Section 9 (Security Considerations) I believe touches on it a 
> little bit, but it seems both incomplete and out of place.
> 
> The problem statement needs to be brough forward in the 
> document (say to section 3 or a new section). The scope of 
> applicability of the solution should also be described up-front).
> 
> I wan't to clarfiy I'm not necessarily against this: I just 
> want the justification to be clearer. The onus should be on 
> this document to clearly demonstrate why what we currently 
> have is broken.
> 
> > -----Original Message-----
> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of 
> > DRAGE, Keith (Keith)
> > Sent: Wednesday, June 04, 2008 02:48
> > To: sip@ietf.org
> > Subject: Re: [Sip] draft-dotson-sip-mutual-auth-02
> > 
> > (As SIP WG cochair)
> > 
> > This has been raised in the SIP group a couple of times, 
> and we have 
> > not yet gained an idea of whether to proceed with it in the 
> SIP WG or 
> > not.
> >  
> > There have been a couple of technical comments raised in 
> the past from 
> > the security experts; my understanding is that these have now been 
> > clarified.
> >  
> > I believe in the past there has been an interested 
> community that says 
> > this is useful. It has also been clarified that 3GPP would 
> like to use 
> > it in their specifications.
> >  
> > So could I ask the WG to look at this document, and formally to 
> > indicate whether they believe some work with this scope should be 
> > progressed as a SIP WG item.
> >  
> > http://www.ietf.org/internet-drafts/draft-dotson-sip-mutual-au
> > th-02.txt
> > 
> > So if you support it please indicate to the SIP WG chairs.
> > 
> > If you have technical concerns (i.e. it should not proceed 
> because it 
> > is technically flawed beyond fixing) then please ideally mail the 
> > list, although the SIP WG chairs will also accept input.
> > 
> > Responses please in 1 calendar week from this date.
> > 
> > Regards
> > 
> > Keith
> > 
> > 
> > ________________________________
> > 
> > 	From: sip-bounces@ietf.org
> > [mailto:sip-bounces@ietf.org] On Behalf Of Stuart Hoggan
> > 	Sent: Friday, May 02, 2008 5:15 PM
> > 	To: sip@ietf.org
> > 	Subject: [Sip] draft-dotson-sip-mutual-auth-02
> > 	
> > 	
> > 
> > 	Folks,
> > 
> > 	 
> > 
> > 	Based on the feedback received in Philadelphia, we have updated 
> > draft-dotson-sip-mutual-auth. The changes clarify the usage of the 
> > Proxy-Authentication-Info header, including support for multiple 
> > values.
> > 
> > 
> > 	 
> > 
> > 	Any further comments or suggestions will be 
> appreciated. It would 
> > also be nice to obtain WG feedback on pursuing this I-D as a WG 
> > document.
> > 
> > 	 
> > 
> > 	Regards,
> > 
> > 	Stuart
> > 
> > 	 
> > 
> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use 
> > sip-implementors@cs.columbia.edu for questions on current sip Use 
> > sipping@ietf.org for new developments on the application of sip
> > 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use 
> sip-implementors@cs.columbia.edu for questions on current sip 
> Use sipping@ietf.org for new developments on the application of sip
> 
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To: sip@ietf.org, discussion@sipforum.org
From: Dale.Worley@comcast.net
In-reply-to: <BAF5F619287DFA4F8BE069A8198CBBB201D578AA@ecnbjmw513.eapac.ericsson.se>
	(yuantao.zhang@ericsson.com)
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Subject: Re: [Sip] What is the port number in "Invite" request-line? Thanks
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   From: "Yuantao Zhang" <yuantao.zhang@ericsson.com>

   INVITE sip:bob@biloxi.com SIP/2.0:5555

This is syntactically invalid.  Do you mean

   INVITE sip:bob@biloxi.com:5555 SIP/2.0

?

Dale
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Subject: Re: [Sip] draft-dotson-sip-mutual-auth-02
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Hi,

I think it would be good to provide a revision, with the addtional informat=
ion, asap. It will be easier to make a decission whether to move the draft =
forward, and it will hopefully reduce the what-is-this-needed-for type of q=
uestions.

Regards,

Christer
 =


-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Suman=
th Channabasappa
Sent: 4. kes=E4kuuta 2008 22:43
To: Francois Audet; DRAGE, Keith (Keith); sip@ietf.org
Subject: Re: [Sip] draft-dotson-sip-mutual-auth-02

Sounds fair, we can provide additional information in a revision (probably =
when the WG decides to pursue this effort?).

Thanks!
- S

-----Original Message-----
From: Francois Audet [mailto:audet@nortel.com]
Sent: Wednesday, June 04, 2008 1:02 PM
To: Sumanth Channabasappa; DRAGE, Keith (Keith); sip@ietf.org
Subject: RE: [Sip] draft-dotson-sip-mutual-auth-02

Ok, thanks.

I would like the begining of the document (say section 3, or a Scope, Appli=
cability statement or similar section) to explains when and why this is use=
ful.

The purpose is really for an implementor to be able by reading this section=
 to be in a position to determine if it's somthing that should he needs to =
worry about or not. =


So, say if I'm developping a UA for an IMS environment versus an Enteprise =
SIP UA, or an Enteprise SIP access to a service provider network, I'll know=
 if this applies to me or not.

Cheers. =


> -----Original Message-----
> From: Sumanth Channabasappa [mailto:sumanth@cablelabs.com]
> Sent: Wednesday, June 04, 2008 11:33
> To: Audet, Francois (SC100:3055); DRAGE, Keith (Keith); sip@ietf.org
> Subject: RE: [Sip] draft-dotson-sip-mutual-auth-02
> =

> Francois,
> =

> Thanks for your thoughts. We can certainly add more details if it is =

> unclear.
> =

> As a clarification, I don't think we are saying that the current =

> mechanism is necessarily broken (which is why it is not an essential =

> correction, as discussed earlier in the WG).
> However, we want to allow for the use of this header in deployments =

> that use Digest (with or without TLS; e.g., when integrity protection =

> is provided by other means) and can benefit from its advantages =

> (mutual auth, nextnonce).
> =

> - S
> =

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =

> Francois Audet
> Sent: Wednesday, June 04, 2008 12:08 PM
> To: DRAGE, Keith (Keith); sip@ietf.org
> Subject: Re: [Sip] draft-dotson-sip-mutual-auth-02
> =

> I do not believe that this document provide enought justification of =

> what is the problem that is being attempted to be solved, and why it =

> isn't already solved by existing mechanism that are widely implemented =

> (such as TLS).
> =

> Section 9 (Security Considerations) I believe touches on it a little =

> bit, but it seems both incomplete and out of place.
> =

> The problem statement needs to be brough forward in the document (say =

> to section 3 or a new section). The scope of applicability of the =

> solution should also be described up-front).
> =

> I wan't to clarfiy I'm not necessarily against this: I just want the =

> justification to be clearer. The onus should be on this document to =

> clearly demonstrate why what we currently have is broken.
> =

> > -----Original Message-----
> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
> Behalf Of
> > DRAGE, Keith (Keith)
> > Sent: Wednesday, June 04, 2008 02:48
> > To: sip@ietf.org
> > Subject: Re: [Sip] draft-dotson-sip-mutual-auth-02
> > =

> > (As SIP WG cochair)
> > =

> > This has been raised in the SIP group a couple of times,
> and we have
> > not yet gained an idea of whether to proceed with it in the
> SIP WG or
> > not.
> >  =

> > There have been a couple of technical comments raised in
> the past from
> > the security experts; my understanding is that these have now been =

> > clarified.
> >  =

> > I believe in the past there has been an interested
> community that says
> > this is useful. It has also been clarified that 3GPP would
> like to use
> > it in their specifications.
> >  =

> > So could I ask the WG to look at this document, and formally to =

> > indicate whether they believe some work with this scope should be =

> > progressed as a SIP WG item.
> >  =

> > http://www.ietf.org/internet-drafts/draft-dotson-sip-mutual-au
> > th-02.txt
> > =

> > So if you support it please indicate to the SIP WG chairs.
> > =

> > If you have technical concerns (i.e. it should not proceed
> because it
> > is technically flawed beyond fixing) then please ideally mail the =

> > list, although the SIP WG chairs will also accept input.
> > =

> > Responses please in 1 calendar week from this date.
> > =

> > Regards
> > =

> > Keith
> > =

> > =

> > ________________________________
> > =

> > 	From: sip-bounces@ietf.org
> > [mailto:sip-bounces@ietf.org] On Behalf Of Stuart Hoggan
> > 	Sent: Friday, May 02, 2008 5:15 PM
> > 	To: sip@ietf.org
> > 	Subject: [Sip] draft-dotson-sip-mutual-auth-02
> > 	=

> > 	=

> > =

> > 	Folks,
> > =

> > 	 =

> > =

> > 	Based on the feedback received in Philadelphia, we have updated =

> > draft-dotson-sip-mutual-auth. The changes clarify the usage of the =

> > Proxy-Authentication-Info header, including support for multiple =

> > values.
> > =

> > =

> > 	 =

> > =

> > 	Any further comments or suggestions will be
> appreciated. It would
> > also be nice to obtain WG feedback on pursuing this I-D as a WG =

> > document.
> > =

> > 	 =

> > =

> > 	Regards,
> > =

> > 	Stuart
> > =

> > 	 =

> > =

> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use =

> > sip-implementors@cs.columbia.edu for questions on current sip Use =

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

> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use =

> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

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


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We found a couple of typos and other nits in draft-ietf-sip- 
outbound-14. The authors are working on a -15 that I expect to be  
"final", so if you have any other changes you want in there, get them  
to the list NOW.

--
Dean

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Hi,
 
One minor editorial issue:
 
Chapter 4.2.2 says:
 
"Re-registrations and single Contact de-registrations use the same instance-id and reg-id values as the corresponding initial registration."
 
Other than the "Contact de-registrations" wording, I can't find any text about explicitly removing flows (using REGISTER with "0" expiration interval). Would it be possible to add some extra words about that?
 
Maybe simply saying:
 
"Re-registrations and single Contact de-registrations (i.e. explicit removal of flows) use the same..." 
 
Regards,
 
Christer

________________________________

From: sip-bounces@ietf.org on behalf of Dean Willis
Sent: Thu 05/06/2008 23:34
To: sip@ietf.org
Subject: [Sip] What's happening with Outbound?




We found a couple of typos and other nits in draft-ietf-sip-
outbound-14. The authors are working on a -15 that I expect to be 
"final", so if you have any other changes you want in there, get them 
to the list NOW.

--
Dean

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


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


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Thread-Topic: [Sip] What is the port number in "Invite" request-line? Thanks
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From: "Yuantao Zhang" <yuantao.zhang@ericsson.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>,
	"Attila Sipos" <attila.sipos@vegastream.com>, <sip@ietf.org>,
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Dear all=20

Thanks for your reply. Please let me ask my question in another way.

=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=3D=3D=3D=3D=3D=3D
UDP:
Src port: 5060
Dst port: 5555        *<--------------
=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=3D=3D=3D=3D=3D=3D
SIP:
INVITE sip:123101@192.168.100.100:5555 SIP/2.0  *<--------------
Via: SIP/2.0/UDP 192.168.100.3:5060;rport;branch=3Dz9hG4bK27195
From: <sip:123102@192.168.100.100>;tag=3D27266
To: <sip:123101@192.168.100.100:5555>
Call-ID: 24658@192.168.100.3
CSeq: 20 INVITE
Contact: <sip:123102@192.168.100.3:5060>
=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=3D=3D=3D=3D=3D=3D

The above "Invite" message is over UDP. In the UDP part (Transport =
layer), the Destination port is 5555. In the SIP part (application =
layer), the port in the "Invite" request-line URL is 5555 too. Both of =
them are same. In which case the port in "Invite" request-line URL is =
different from the Destination port in lower layer UDP? Furthermore, if =
they are always same why we include redundant information (the port =
number) in "Invite" request-line URL? What happens if the port in =
"Invite" request-line URL is different from the Destination port in =
lower layer UDP?=20

Thank you very much

Best Regards
Yuantao
-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
Sent: 2008=C4=EA6=D4=C24=C8=D5 20:45
To: Attila Sipos
Cc: Yuantao Zhang; sip@ietf.org; discussion@sipforum.org
Subject: Re: [Sip] What is the port number in "Invite" request-line? =
Thanks



Attila Sipos wrote:
> =20
> It is possible to have two instances of a UA (for example) on one =
unit.
>=20
> So using the port is theoretically one way to differentiate between=20
> instances.
> =20
> It's not useless.  If you get a Contact header with a port then you =
have
> to send to that port (discounting NAT traversal scenarios) otherwise=20
> everything will break.
> =20
>> >Or is the port number is which the UAS is listening to? How the UAC=20
> knows it is 5555 if so?
> =20
> A UAC might not know but a proxy/registrar might - for example if a =
user=20
> was registered with
> the Contact as port 5555, then requests to that user will get =
forwarded=20
> to port 5555.
> Another user using, for example, port 5556 will register using a =
Contact=20
> with 5556
> and so will receive requests to that port.

Right. This illustrates a good point:

In general a UAC should not be expected to know "what port a UAS listens =

on". All A UAC should be doing is sending requests to URIs that it has=20
been given. If the URI it is given contains a port number, then it=20
should honor that.

This gets bent out of shape a bit by numeric dialing, where the UAC is=20
only given a dial string and must derive a URI from it. The "standard"=20
way to do this is via ENUM, where effectively the numeric address is=20
looked up in a DB to obtain a URI. In that case again the UAC has been=20
"given" the URI, which may have a port or not.

In the non-standard but common case where the UAC *constructs* a URI=20
from a dial string, the UAC must still be *told* what to use for the=20
remainder of the URI.

	Paul


From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =
Yuantao Zhang
Sent: 04 June 2008 11:10
To: sip@ietf.org; discussion@sipforum.org
Subject: [Sip] What is the port number in "Invite" request-line? Thanks
Dear all
INVITE sip:bob@biloxi.com SIP/2.0:5555
INVITE sip:bob@biloxi.com SIP/2.0
Above are 2 =A1=B0Invite=A1=B1 request-line examples. In example one, it =
indicates a port number, 5555. What does the port number mean?=20
Is it the port number of SIP proxy? Why we include port number here? The =
SIP proxy(biloxi.com) application layer who receives this "Invite" =
should receive this via port 5555 from lowe layer(UDP). Therefore, The =
SIP proxy(biloxi.com) application layer already know the port is 5555. =
Actualy, the SIP proxy application layer keeps listening port 5555.=20
The second =A1=B0Invite=A1=B1 request-line example is from RFC3261 and =
does not include port number. Can I say the port number is useless? So =
why example one includes a useless part in =A1=B0Invite=A1=B1 =
request-line URL?=20
Or is the port number is which the UAS is listening to? How the UAC =
knows it is 5555 if so?=20
Any standard on this port number issue? Thanks.=20
Best regards
Steven


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<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
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FACE=3D"Courier New">w</FONT><FONT SIZE=3D2 FACE=3D"Courier =
New">a</FONT><FONT SIZE=3D2 FACE=3D"Courier New">y</FONT><FONT SIZE=3D2 =
FACE=3D"Courier New">.</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">=3D</FONT><FONT SIZE=3D2 FACE=3D"Courier =
New">=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=
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<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN LANG=3D"fr"><FONT =
SIZE=3D2 FACE=3D"Courier New">UDP:</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"fr"><FONT SIZE=3D2 FACE=3D"Courier New">Src =
port: 5060</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"fr"><FONT SIZE=3D2 FACE=3D"Courier New">Dst =
port:</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN LANG=3D"fr"> <FONT =
COLOR=3D"#FF0000" SIZE=3D2 FACE=3D"Courier New">5555</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"fr"><FONT COLOR=3D"#FF0000" SIZE=3D2 =
FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"fr"> <FONT COLOR=3D"#FF0000" =
SIZE=3D2 FACE=3D"Courier New">*</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"fr"><FONT FACE=3D"Wingdings" =
SIZE=3D2>&#223;</FONT></SPAN><SPAN LANG=3D"fr"></SPAN><SPAN =
LANG=3D"fr"></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN LANG=3D"fr"><FONT =
COLOR=3D"#FF0000" SIZE=3D2 FACE=3D"Courier New">----</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"fr"><FONT COLOR=3D"#FF0000" SIZE=3D2 =
FACE=3D"Courier New">--------</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"fr"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">=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=3D=3D=3D=3D=3D=3D</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">SIP:</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">INVITE sip:123101@192.168.100.100:</FONT></SPAN><SPAN =
LANG=3D"en-us"><FONT COLOR=3D"#FF0000" SIZE=3D2 FACE=3D"Courier =
New">5555</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 =
FACE=3D"Courier New"> SIP/2.0</FONT></SPAN><SPAN LANG=3D"en-us"><FONT =
SIZE=3D2 FACE=3D"Courier New">&nbsp;</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"fr"> <FONT COLOR=3D"#FF0000" =
SIZE=3D2 FACE=3D"Courier New">*</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"fr"><FONT FACE=3D"Wingdings" =
SIZE=3D2>&#223;</FONT></SPAN><SPAN LANG=3D"fr"></SPAN><SPAN =
LANG=3D"fr"></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN LANG=3D"fr"><FONT =
COLOR=3D"#FF0000" SIZE=3D2 FACE=3D"Courier =
New">------------</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Via: SIP/2.0/UDP =
192.168.100.3:5060;rport;branch=3Dz9hG4bK27195</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">From: &lt;s</FONT><FONT SIZE=3D2 FACE=3D"Courier =
New">ip:123102@192.168.100.100&gt;;tag=3D27266</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">To: &lt;sip:123101@192.168.100.100:</FONT><FONT SIZE=3D2 =
FACE=3D"Courier New">5555</FONT><FONT SIZE=3D2 FACE=3D"Courier =
New">&gt;</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Call-ID: 24658@192.168.100.3</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">CSeq: 20 INVITE</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Contact: &lt;sip:123102@192.168.100.3:5060&gt;</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">=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=3D=3D=3D=3D=3D=3D</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">T</FONT><FONT SIZE=3D2 FACE=3D"Courier New">he above =
&quot;</FONT><FONT SIZE=3D2 FACE=3D"Courier New">In</FONT><FONT SIZE=3D2 =
FACE=3D"Courier New">vite</FONT><FONT SIZE=3D2 FACE=3D"Courier =
New">&quot;</FONT><FONT SIZE=3D2 FACE=3D"Courier New"> =
message</FONT><FONT SIZE=3D2 FACE=3D"Courier New"> is =
over</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT SIZE=3D2 FACE=3D"Courier =
New">UDP.</FONT> <FONT SIZE=3D2 FACE=3D"Courier New">In the =
UDP</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT SIZE=3D2 FACE=3D"Courier =
New">part (</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 =
FACE=3D"Arial">Transport</FONT> <FONT SIZE=3D2 =
FACE=3D"Arial">layer</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 =
FACE=3D"Courier New">)</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 =
FACE=3D"Courier New">, the Destination port is 555</FONT></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New">5.</FONT> <FONT =
SIZE=3D2 FACE=3D"Courier New">In the SIP</FONT></SPAN><SPAN =
LANG=3D"en-us"> <FONT SIZE=3D2 FACE=3D"Courier New">part =
(</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">application layer</FONT><FONT SIZE=3D2 FACE=3D"Courier =
New">)</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">,</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New"></FONT></SPAN><SPAN LANG=3D"en-us"> <FONT SIZE=3D2 FACE=3D"Courier =
New">t</FONT><FONT SIZE=3D2 FACE=3D"Courier New">he</FONT></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New"> port</FONT> <FONT =
SIZE=3D2 FACE=3D"Courier New">in the</FONT></SPAN><SPAN LANG=3D"en-us"> =
<FONT SIZE=3D2 FACE=3D"Courier New">&quot;</FONT><FONT SIZE=3D2 =
FACE=3D"Courier New">In</FONT><FONT SIZE=3D2 FACE=3D"Courier =
New">v</FONT><FONT SIZE=3D2 FACE=3D"Courier New">ite</FONT><FONT =
SIZE=3D2 FACE=3D"Courier New">&quot;</FONT> <FONT SIZE=3D2 =
FACE=3D"Courier New">request</FONT><FONT SIZE=3D2 FACE=3D"Courier =
New">-</FONT><FONT SIZE=3D2 FACE=3D"Courier New">lin</FONT><FONT =
SIZE=3D2 FACE=3D"Courier New">e</FONT></SPAN><SPAN LANG=3D"en-us"><FONT =
SIZE=3D2 FACE=3D"Courier New"> UR</FONT><FONT SIZE=3D2 FACE=3D"Courier =
New">L</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT SIZE=3D2 FACE=3D"Courier =
New">is</FONT> <FONT SIZE=3D2 FACE=3D"Courier New">5555</FONT><FONT =
SIZE=3D2 FACE=3D"Courier New"> too</FONT></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New">.</FONT> <FONT =
SIZE=3D2 FACE=3D"Courier New">Both of them are</FONT></SPAN><SPAN =
LANG=3D"en-us"> <FONT SIZE=3D2 FACE=3D"Courier =
New">same</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 =
FACE=3D"Courier New">. In which case</FONT></SPAN><SPAN LANG=3D"en-us"> =
<FONT SIZE=3D2 FACE=3D"Courier New">the port in</FONT></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New"></FONT></SPAN><SPAN =
LANG=3D"en-us"> <FONT SIZE=3D2 FACE=3D"Courier New">&quot;</FONT><FONT =
SIZE=3D2 FACE=3D"Courier New">In</FONT><FONT SIZE=3D2 FACE=3D"Courier =
New">v</FONT><FONT SIZE=3D2 FACE=3D"Courier New">ite</FONT><FONT =
SIZE=3D2 FACE=3D"Courier New">&quot;</FONT> <FONT SIZE=3D2 =
FACE=3D"Courier New">request</FONT><FONT SIZE=3D2 FACE=3D"Courier =
New">-</FONT><FONT SIZE=3D2 FACE=3D"Courier New">lin</FONT><FONT =
SIZE=3D2 FACE=3D"Courier New">e</FONT></SPAN><SPAN LANG=3D"en-us"><FONT =
SIZE=3D2 FACE=3D"Courier New"> URL is different from the Destination =
port</FONT> <FONT SIZE=3D2 FACE=3D"Courier New">in</FONT></SPAN><SPAN =
LANG=3D"en-us"> <FONT SIZE=3D2 FACE=3D"Courier New">l</FONT><FONT =
SIZE=3D2 FACE=3D"Courier New">ower</FONT></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New"></FONT> <FONT =
SIZE=3D2 FACE=3D"Courier New">layer</FONT></SPAN><SPAN LANG=3D"en-us"> =
<FONT SIZE=3D2 FACE=3D"Courier New">UDP?</FONT></SPAN><SPAN =
LANG=3D"en-us"> <FONT SIZE=3D2 FACE=3D"Courier New">F</FONT><FONT =
SIZE=3D2 FACE=3D"Courier New">urthermore, if th</FONT><FONT SIZE=3D2 =
FACE=3D"Courier New">ey are always same</FONT></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New"> why we =
include</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT SIZE=3D2 =
FACE=3D"Courier New">redundan</FONT></SPAN><SPAN LANG=3D"en-us"><FONT =
SIZE=3D2 FACE=3D"Courier New">t</FONT><FONT SIZE=3D2 FACE=3D"Courier =
New"> information</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 =
FACE=3D"Courier New"></FONT> <FONT SIZE=3D2 FACE=3D"Courier =
New">(</FONT><FONT SIZE=3D2 FACE=3D"Courier New">the port =
number</FONT><FONT SIZE=3D2 FACE=3D"Courier New">)</FONT></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New"></FONT> <FONT =
SIZE=3D2 FACE=3D"Courier New">in</FONT></SPAN><SPAN LANG=3D"en-us"> =
<FONT SIZE=3D2 FACE=3D"Courier New">&quot;</FONT><FONT SIZE=3D2 =
FACE=3D"Courier New">In</FONT><FONT SIZE=3D2 FACE=3D"Courier =
New">v</FONT><FONT SIZE=3D2 FACE=3D"Courier New">ite</FONT><FONT =
SIZE=3D2 FACE=3D"Courier New">&quot;</FONT> <FONT SIZE=3D2 =
FACE=3D"Courier New">request</FONT><FONT SIZE=3D2 FACE=3D"Courier =
New">-</FONT><FONT SIZE=3D2 FACE=3D"Courier New">lin</FONT><FONT =
SIZE=3D2 FACE=3D"Courier New">e</FONT></SPAN><SPAN LANG=3D"en-us"><FONT =
SIZE=3D2 FACE=3D"Courier New"> UR</FONT><FONT SIZE=3D2 FACE=3D"Courier =
New">L</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">?</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New"></FONT></SPAN><SPAN LANG=3D"en-us"> <FONT SIZE=3D2 FACE=3D"Courier =
New">What</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT SIZE=3D2 =
FACE=3D"Courier New">happen</FONT></SPAN><SPAN LANG=3D"en-us"><FONT =
SIZE=3D2 FACE=3D"Courier New">s if</FONT><FONT SIZE=3D2 FACE=3D"Courier =
New"></FONT></SPAN><SPAN LANG=3D"en-us"> <FONT SIZE=3D2 FACE=3D"Courier =
New">the port in</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 =
FACE=3D"Courier New"></FONT></SPAN><SPAN LANG=3D"en-us"> <FONT SIZE=3D2 =
FACE=3D"Courier New">&quot;</FONT><FONT SIZE=3D2 FACE=3D"Courier =
New">In</FONT><FONT SIZE=3D2 FACE=3D"Courier New">v</FONT><FONT SIZE=3D2 =
FACE=3D"Courier New">ite</FONT><FONT SIZE=3D2 FACE=3D"Courier =
New">&quot;</FONT> <FONT SIZE=3D2 FACE=3D"Courier =
New">request</FONT><FONT SIZE=3D2 FACE=3D"Courier New">-</FONT><FONT =
SIZE=3D2 FACE=3D"Courier New">lin</FONT><FONT SIZE=3D2 FACE=3D"Courier =
New">e</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New"> URL is different from the Destination port</FONT> <FONT SIZE=3D2 =
FACE=3D"Courier New">in</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT =
SIZE=3D2 FACE=3D"Courier New">l</FONT><FONT SIZE=3D2 FACE=3D"Courier =
New">ower</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 =
FACE=3D"Courier New"></FONT> <FONT SIZE=3D2 FACE=3D"Courier =
New">layer</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT SIZE=3D2 =
FACE=3D"Courier New">UD</FONT><FONT SIZE=3D2 FACE=3D"Courier =
New">P</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">?</FONT></SPAN><SPAN LANG=3D"en-us"> </SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Thank you very much</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Best Regards</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Yuantao</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">-----Original Message-----</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">From: Paul Kyzivat [<A =
HREF=3D"mailto:pkyzivat@cisco.com">mailto:pkyzivat@cisco.com</A>] =
</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Sent: 2008</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 =
FACE=3D"=CB=CE=CC=E5">=C4=EA</FONT></SPAN><SPAN LANG=3D"en-us"><FONT =
SIZE=3D2 FACE=3D"Courier New">6</FONT></SPAN><SPAN LANG=3D"en-us"><FONT =
SIZE=3D2 FACE=3D"=CB=CE=CC=E5">=D4=C2</FONT></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New">4</FONT></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D2 =
FACE=3D"=CB=CE=CC=E5">=C8=D5</FONT></SPAN><SPAN LANG=3D"en-us"><FONT =
SIZE=3D2 FACE=3D"Courier New"> 20:45</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">To: Attila Sipos</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Cc: Yuantao Zhang; sip@ietf.org; =
discussion@sipforum.org</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Subject: Re: [Sip] What is the port number in &quot;Invite&quot; =
request-line? Thanks</FONT></SPAN></P>
<BR>
<BR>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Attila Sipos wrote:</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&gt;&nbsp; </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&gt; It is possible to have two instances of a UA (for example) on =
one unit.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&gt; </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&gt; So using the port</FONT><FONT SIZE=3D2 FACE=3D"Courier New"> =
is theoretically one way to differentiate between </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&gt; instances.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&gt;&nbsp; </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&gt; It's not useless.&nbsp; If you get a Contact header with a =
port then you have</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&gt; to send to that port (discounting NAT traversal scenarios) =
otherwise </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&gt; everything will break.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&gt;&nbsp; </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&gt;&gt; &gt;Or is t</FONT><FONT SIZE=3D2 FACE=3D"Courier New">he =
port number is which the UAS is listening to? How the UAC =
</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&gt; knows it is 5555 if so?</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&gt;&nbsp; </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&gt; A UAC might not know but a proxy/registrar might - for example =
if a user </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&gt; was registered with</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&gt; the Contact as port 5555, then requests to that user will get =
forwarded </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&gt; to port 5555.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&gt; Another user using, for example, port 5556 will register using =
a Contact </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&gt; with 5556</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&gt; and so will receive requests to that port.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Right. This illustrates a good</FONT> <FONT SIZE=3D2 =
FACE=3D"Courier New">point:</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New">In =
general a UAC should not be expected to know &quot;what port a UAS =
listens </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">on&quot;. All A UAC should be doing is sending requests to URIs =
that it has </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">been given. If the URI it is given contains a port number, then it =
</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">should honor that.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">This gets bent</FONT><FONT SIZE=3D2 FACE=3D"Courier New"> out of =
shape a bit by numeric dialing, where the UAC is </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">only given a dial string and must derive a URI from it. The =
&quot;standard&quot; </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">way to do this is via ENUM, where effectively the numeric address =
is </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">looked up in a DB to obtain a URI. In that case again =
th</FONT><FONT SIZE=3D2 FACE=3D"Courier New">e UAC has been =
</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&quot;given&quot; the URI, which may have a port or =
not.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier New">In =
the non-standard but common case where the UAC *constructs* a URI =
</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">from a dial string, the UAC must still be *told* what to use for =
the </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">remainder of the URI.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN =
LANG=3D"en-us">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2 =
FACE=3D"Courier New">Pau</FONT><FONT SIZE=3D2 FACE=3D"Courier =
New">l</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"><B></B></SPAN><SPAN LANG=3D"en-us"><B><FONT SIZE=3D2 =
FACE=3D"Tahoma">From:</FONT></B></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Tahoma"> sip-bounces@ietf.org [<A =
HREF=3D"mailto:sip-bounces@ietf.org">mailto:sip-bounces@ietf.org</A>]</FO=
NT></SPAN><SPAN LANG=3D"en-us"><B></B></SPAN><SPAN LANG=3D"en-us"><B> =
<FONT SIZE=3D2 FACE=3D"Tahoma">On Behalf Of</FONT></B></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"> <FONT SIZE=3D2 =
FACE=3D"Tahoma">Yuantao Zhang<BR>
</FONT></SPAN><SPAN LANG=3D"en-us"><B></B></SPAN><SPAN =
LANG=3D"en-us"><B><FONT SIZE=3D2 =
FACE=3D"Tahoma">Sent:</FONT></B></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Tahoma"> 04 June 2008 11:10<BR>
</FONT></SPAN><SPAN LANG=3D"en-us"><B></B></SPAN><SPAN =
LANG=3D"en-us"><B><FONT SIZE=3D2 =
FACE=3D"Tahoma">To:</FONT></B></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Tahoma"> sip@ietf.org; =
discussion@sipforum.org<BR>
</FONT></SPAN><SPAN LANG=3D"en-us"><B></B></SPAN><SPAN =
LANG=3D"en-us"><B><FONT SIZE=3D2 =
FACE=3D"Tahoma">Subject:</FONT></B></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 =
FACE=3D"Tahoma"> [Sip] What is the port number in &quot;Invite&quot; =
request-line? Thanks</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT =
COLOR=3D"#000080" SIZE=3D2 FACE=3D"Arial">Dear al</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT COLOR=3D"#000080" =
SIZE=3D2 FACE=3D"Arial">l</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT =
COLOR=3D"#000080" SIZE=3D2 FACE=3D"Arial">INVITE =
sip:bob@biloxi</FONT><FONT COLOR=3D"#000080" SIZE=3D2 =
FACE=3D"Arial">.com SIP/2.0:555</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT COLOR=3D"#000080" =
SIZE=3D2 FACE=3D"Arial">5</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT =
COLOR=3D"#000080" SIZE=3D2 FACE=3D"Arial">INVITE sip:bob@biloxi.com =
SIP/2.</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"><FONT COLOR=3D"#000080" SIZE=3D2 =
FACE=3D"Arial">0</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT =
COLOR=3D"#000080" SIZE=3D2 FACE=3D"Arial">Above are 2 =A1=B0</FONT><FONT =
COLOR=3D"#000080" SIZE=3D2 FACE=3D"Arial">Invite=A1=B1</FONT><FONT =
COLOR=3D"#000080" SIZE=3D2 FACE=3D"Arial"> request-line examples. In =
example one, it indicates a port number, 5555. What does the port number =
mean?</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"><FONT COLOR=3D"#000080" SIZE=3D2 =
FACE=3D"Arial"></FONT></SPAN><SPAN LANG=3D"en-us"> </SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT =
COLOR=3D"#000080" SIZE=3D2 FACE=3D"Arial">Is it the port number of SIP =
proxy? Why we include port number here? The SIP</FONT> <FONT =
COLOR=3D"#000080" SIZE=3D2 FACE=3D"Arial">proxy(biloxi.com) application =
layer who receives this &quot;Invite&quot; should receive this via port =
5555 from lowe layer(UDP).</FONT> <FONT COLOR=3D"#000080" SIZE=3D2 =
FACE=3D"Arial">T</FONT><FONT COLOR=3D"#000080" SIZE=3D2 =
FACE=3D"Arial">herefo</FONT><FONT COLOR=3D"#000080" SIZE=3D2 =
FACE=3D"Arial">r</FONT><FONT COLOR=3D"#000080" SIZE=3D2 =
FACE=3D"Arial">e, The SIP proxy(biloxi.com) application layer already =
know the port is 5555. Actualy, the SIP proxy application layer keeps =
list</FONT><FONT COLOR=3D"#000080" SIZE=3D2 FACE=3D"Arial">ening port =
5555.</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"><FONT COLOR=3D"#000080" SIZE=3D2 =
FACE=3D"Arial"></FONT></SPAN><SPAN LANG=3D"en-us"> </SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT =
COLOR=3D"#000080" SIZE=3D2 FACE=3D"Arial">The second =A1=B0</FONT><FONT =
COLOR=3D"#000080" SIZE=3D2 FACE=3D"Arial">Invite=A1=B1</FONT><FONT =
COLOR=3D"#000080" SIZE=3D2 FACE=3D"Arial"> request-line example is from =
RFC3261 and does not include port number. Can I say the port number is =
useless? So why example one includes a useless part in =
=A1=B0</FONT><FONT COLOR=3D"#000080" SIZE=3D2 =
FACE=3D"Arial">Invite=A1=B1</FONT><FONT COLOR=3D"#000080" SIZE=3D2 =
FACE=3D"Arial"> request-line URL?</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT COLOR=3D"#000080" =
SIZE=3D2 FACE=3D"Arial"></FONT></SPAN><SPAN LANG=3D"en-us"> </SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT =
COLOR=3D"#000080" SIZE=3D2 FACE=3D"Arial">Or is the port number</FONT> =
<FONT COLOR=3D"#000080" SIZE=3D2 FACE=3D"Arial">is which the UAS is =
listening to? How the UAC knows it is 5555 if so?</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT COLOR=3D"#000080" =
SIZE=3D2 FACE=3D"Arial"></FONT></SPAN><SPAN LANG=3D"en-us"> </SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT =
COLOR=3D"#000080" SIZE=3D2 FACE=3D"Arial">Any standard on this port =
number issue? Thanks.</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"><FONT COLOR=3D"#000080" SIZE=3D2 =
FACE=3D"Arial"></FONT></SPAN><SPAN LANG=3D"en-us"> </SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT =
COLOR=3D"#000080" SIZE=3D2 FACE=3D"Arial">Best regard</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT COLOR=3D"#000080" =
SIZE=3D2 FACE=3D"Arial">s</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT =
COLOR=3D"#000080" SIZE=3D2 FACE=3D"Arial">Steve</FONT><FONT =
COLOR=3D"#000080" SIZE=3D2 FACE=3D"Arial">n</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN></P>

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From sip-bounces@ietf.org  Thu Jun  5 22:46:10 2008
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From: atul garg <shivalk_80@yahoo.co.in>
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Subject: [Sip] Call ID doubt in SIP and PTT
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Hello,

I have two queries  -
 
1) When we say there is a SIP Conference ie more than two parties are in one call session then how call-id are maintained that is Call -id remains same or changes ,more precisely if SIP Client-A want to invite Client-B and C for call ,it sends a SIP INvite to SIP server-A and ask it to call B and C. Now when SIP server will send two Invites ,one to B and second to c will all these Call IDs be different or same.

2) Same is for Push To Talk(PTT) , when there is a chat session established between multiple clients , are the Call id s same or different or else how one can correlate different messages belong to one session ?/ is there any session id or some other unique field present ??

Please also suggest if anyone have some study documnt regarding this.

Thanx
Shiva


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From sip-bounces@ietf.org  Fri Jun  6 02:30:32 2008
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From: cco <cristian.constantin@iptel.org>
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Subject: [Sip] aka authentication and sip (rfc 3310)
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hi!

in either of the signalling flow examples given in the rfc 3310 (4. Example
Digest AKA Operation), the first non-auth REGISTER does not contain an
"Authorization" header. also the rfc does not seem to enforce the usage
of an "Authorization" header right on from the first register.

now, considering that:

1. an correct AKA challenge is user dependent.
2. in order to generate the correct AKA challenge the proxy may need
information about both:
a. the aor (From/To headers of the REGISTER) and
b. the authentication username ("username" parameter of the "Authorization" 
header)

the question is how should the proxy generate a correct AKA challenge when
the "Authorization" header is not present in the first (non-auth.)
REGISTER?

bye now!
cristian
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Subject: Re: [Sip] What is the port number in "Invite" request-line? Thanks
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Date: Fri, 06 Jun 2008 16:49:59 +0200
From: Dragos Vingarzan <Dragos.Vingarzan@fokus.fraunhofer.de>
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To: cco <cristian.constantin@iptel.org>
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Subject: Re: [Sip] aka authentication and sip (rfc 3310)
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I also find this quite annoying. The 3GPP standards indicate that the UE 
"shall" add that initial header, so if you are in the 3GPP IMS scope, it 
works.

After implementing AKA and using it for quite some time, I would say 
that the presence of an Authorization header in the first request 
indicates that the client knows how to do AKA. If you don't see one, 
don't bother sending an AKA challenge. But this is just an experience 
fact, so to say, and not a written fact.

And I would add that I don't think that there is another way of 
auto-detecting the username either. The best example is when I have 2 
phones with 2 ISIMs, but they share the same AOR. The requests would be 
identical for both, yet the challenge must be generated accordingly for 
each of them.

I hope we get a more authoritative answer though.... Anyone?

Cheers,
-Dragos


cco wrote:
> hi!
>
> in either of the signalling flow examples given in the rfc 3310 (4. Example
> Digest AKA Operation), the first non-auth REGISTER does not contain an
> "Authorization" header. also the rfc does not seem to enforce the usage
> of an "Authorization" header right on from the first register.
>
> now, considering that:
>
> 1. an correct AKA challenge is user dependent.
> 2. in order to generate the correct AKA challenge the proxy may need
> information about both:
> a. the aor (From/To headers of the REGISTER) and
> b. the authentication username ("username" parameter of the "Authorization" 
> header)
>
> the question is how should the proxy generate a correct AKA challenge when
> the "Authorization" header is not present in the first (non-auth.)
> REGISTER?
>
> bye now!
> cristian
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>
>   
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


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	"Dean Willis" <dean.willis@softarmor.com>, <sip@ietf.org>
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I propose that we use the exact terminology of RFC 3261 (10.2.1 -adding-,
10.2.3 -removing- and 10.2.4 -refreshing).

So it would look like this:

	Registrations for refreshing a binding and for removing 
	a binding use the same instance-id and reg-id values as the 
	corresponding initial registration where the binding
	was added.



> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of Christer Holmberg
> Sent: Thursday, June 05, 2008 20:04
> To: Dean Willis; sip@ietf.org
> Subject: Re: [Sip] What's happening with Outbound?
> 
> Hi,
>  
> One minor editorial issue:
>  
> Chapter 4.2.2 says:
>  
> "Re-registrations and single Contact de-registrations use the 
> same instance-id and reg-id values as the corresponding 
> initial registration."
>  
> Other than the "Contact de-registrations" wording, I can't 
> find any text about explicitly removing flows (using REGISTER 
> with "0" expiration interval). Would it be possible to add 
> some extra words about that?
>  
> Maybe simply saying:
>  
> "Re-registrations and single Contact de-registrations (i.e. 
> explicit removal of flows) use the same..." 
>  
> Regards,
>  
> Christer
> 
> ________________________________
> 
> From: sip-bounces@ietf.org on behalf of Dean Willis
> Sent: Thu 05/06/2008 23:34
> To: sip@ietf.org
> Subject: [Sip] What's happening with Outbound?
> 
> 
> 
> 
> We found a couple of typos and other nits in draft-ietf-sip- 
> outbound-14. The authors are working on a -15 that I expect 
> to be "final", so if you have any other changes you want in 
> there, get them to the list NOW.
> 
> --
> Dean
> 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use 
> sip-implementors@cs.columbia.edu for questions on current sip 
> Use sipping@ietf.org for new developments on the application of sip
> 
> 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use 
> sip-implementors@cs.columbia.edu for questions on current sip 
> Use sipping@ietf.org for new developments on the application of sip
> 
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From sip-bounces@ietf.org  Fri Jun  6 11:18:30 2008
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Hi,
 
If "removing a binding" = "removing a flow", then I guess it's ok.
 
REgards,
 
Christer

________________________________

From: Francois Audet [mailto:audet@nortel.com]
Sent: Fri 06/06/2008 18:39
To: Christer Holmberg; Dean Willis; sip@ietf.org
Cc: Cullen Jennings; Rohan Mahy
Subject: RE: [Sip] What's happening with Outbound?



I propose that we use the exact terminology of RFC 3261 (10.2.1 -adding-,
10.2.3 -removing- and 10.2.4 -refreshing).

So it would look like this:

        Registrations for refreshing a binding and for removing
        a binding use the same instance-id and reg-id values as the
        corresponding initial registration where the binding
        was added.



> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
> Behalf Of Christer Holmberg
> Sent: Thursday, June 05, 2008 20:04
> To: Dean Willis; sip@ietf.org
> Subject: Re: [Sip] What's happening with Outbound?
>
> Hi,
> 
> One minor editorial issue:
> 
> Chapter 4.2.2 says:
> 
> "Re-registrations and single Contact de-registrations use the
> same instance-id and reg-id values as the corresponding
> initial registration."
> 
> Other than the "Contact de-registrations" wording, I can't
> find any text about explicitly removing flows (using REGISTER
> with "0" expiration interval). Would it be possible to add
> some extra words about that?
> 
> Maybe simply saying:
> 
> "Re-registrations and single Contact de-registrations (i.e.
> explicit removal of flows) use the same..."
> 
> Regards,
> 
> Christer
>
> ________________________________
>
> From: sip-bounces@ietf.org on behalf of Dean Willis
> Sent: Thu 05/06/2008 23:34
> To: sip@ietf.org
> Subject: [Sip] What's happening with Outbound?
>
>
>
>
> We found a couple of typos and other nits in draft-ietf-sip-
> outbound-14. The authors are working on a -15 that I expect
> to be "final", so if you have any other changes you want in
> there, get them to the list NOW.
>
> --
> Dean
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use
> sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use
> sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>


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


From c.jordan@aldridge-jordan.com  Fri Jun  6 23:38:30 2008
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From sip-bounces@ietf.org  Fri Jun  6 23:59:05 2008
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On Jun 6, 2008, at 12:45 AM, atul garg wrote:

> Hello,
>
> I have two queries  -
>
> 1) When we say there is a SIP Conference ie more than two parties  
> are in one call session then how call-id are maintained that is Call  
> -id remains same or changes ,more precisely if SIP Client-A want to  
> invite Client-B and C for call ,it sends a SIP INvite to SIP server- 
> A and ask it to call B and C. Now when SIP server will send two  
> Invites ,one to B and second to c will all these Call IDs be  
> different or same.
>
> 2) Same is for Push To Talk(PTT) , when there is a chat session  
> established between multiple clients , are the Call id s same or  
> different or else how one can correlate different messages belong to  
> one session ?/ is there any session id or some other unique field  
> present ??
>
> Please also suggest if anyone have some study documnt regarding this.
>

Most SIP "conferences" are implemented by conference bridges, which  
are back-to-back user agents. What do they do with call IDs? Whatever  
they want to -- there behavior really isn't specified by IETF.

usually, each "leg" of the bridge will be a separate call with a  
distinct call ID. This makes life much easier.

As I understand OMA POC, this is what happens there -- but you'd have  
to read their spec to be sure. Each "POC controller" (they call it a  
"participating function", IIRC) is basically a B2BUA. So even a  
minimal 2 party POC call has three different call legs, all of which  
probably have (or at least probably should have) different call-IDs.

--
Dean
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From c.glaeser@cglaeser.de  Sun Jun  8 10:34:28 2008
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From sip-bounces@ietf.org  Sun Jun  8 13:08:38 2008
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Date: Sun, 8 Jun 2008 22:08:46 +0200
From: cco <cristian.constantin@iptel.org>
To: Dragos Vingarzan <Dragos.Vingarzan@fokus.fraunhofer.de>
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Subject: Re: [Sip] aka authentication and sip (rfc 3310)
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On Fri, Jun 06, 2008 at 04:49:59PM +0200, Dragos Vingarzan wrote:
> I also find this quite annoying. The 3GPP standards indicate that the UE 
> "shall" add that initial header, so if you are in the 3GPP IMS scope, it 
> works.
> 
> After implementing AKA and using it for quite some time, I would say 
> that the presence of an Authorization header in the first request 
> indicates that the client knows how to do AKA. If you don't see one, 
> don't bother sending an AKA challenge. But this is just an experience 
> fact, so to say, and not a written fact.
> 
> And I would add that I don't think that there is another way of 
> auto-detecting the username either. The best example is when I have 2 
> phones with 2 ISIMs, but they share the same AOR. The requests would be 
> identical for both, yet the challenge must be generated accordingly for 
> each of them.

cristian: hi! dragos, I am refering strictly to the rfc3310 and I am
trying to understand how the call flows given as an example there would
function in the most general case, that is:

1. the AKA key is shared btw. an AuC and a certain authorization username
2. the sip proxy does not have yet the authentication vector and needs to get 
it somehow from the AuC.

now, the call flows given as an example would function when the
challenge (nonce) does not explicitly depend on the user which is
actually trying to register itself, being generated
(periodically) by the sip proxy itself; that is they are o.k. for an
HTTP digest MD5 auth. as spec. in rfc 3261/rfc 2617. but they do NOT seem to 
function in the real world when AKA is used, because:

1. the challenge is user specific.
2. the "network" authenticates itself to the UE as well in case of AKA.

but then again my interpretation of the rfc3310/call flows given as an
example there may be wrong. that is why I am asking for an oppinion on
this mailing list.

thanks.

bye now!
cristian
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From ca01@51translation.com  Sun Jun  8 17:26:11 2008
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Thread-Topic: [Sip] What is the port number in "Invite" request-line? Thanks
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From: "Yuantao Zhang" <yuantao.zhang@ericsson.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>, <sip@ietf.org>,
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Subject: Re: [Sip] What is the port number in "Invite" request-line? Thanks
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------------------     -------------------         -------------------
|UA1		  |	|SIP proxy	    |	      |UA2		  |
|IP:192.168.100.1|-----|IP: 192.168.100.2 |--------|IP: 192.168.100.3 |
|Port: 1111	  |	|Port: 2222        |	      |Port: 3333        |
------------------	--------------------         -------------------

If UA1 makes call to UA2, UA1 should send the following message to the =
SIP proxy:

=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=3D=3D=3D=3D=3D=3D
IP:=20
Src: 192.168.100.1
Dst: 192.168.100.2
=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=3D=3D=3D=3D=3D=3D
UDP:
Src port: 1111
Dst port: 2222        						   !
=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=3D=3D=3D=3D=3D=3D
SIP:
INVITE sip:UA2@192.168.100.2:2222 SIP/2.0  			   !
Via: SIP/2.0/UDP 192.168.100.1:1111;
=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=3D=3D=3D=3D=3D=3D


Then SIP proxy sends the following message to UA2:

=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=3D=3D=3D=3D=3D=3D
IP:=20
Src: 192.168.100.2
Dst: 192.168.100.3
=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=3D=3D=3D=3D=3D=3D
UDP:
Src port: 2222
Dst port: 3333        						   !
=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=3D=3D=3D=3D=3D=3D
SIP:
INVITE sip:UA2@192.168.100.2:3333 SIP/2.0  			   !
Via: SIP/2.0/UDP 192.168.100.2:2222;
Via: SIP/2.0/UDP 192.168.100.1:1111;
=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=3D=3D=3D=3D=3D=3D

UAC(UA1) does not know the port of UAS(UA2). All the UAC needs to know =
is the proxy's port. And the proxy knows all the ports of UAs under it =
(UAs tell proxy when register). Therefore, the proxy does not care the =
port number in the received "Invite" request-linen (because the port =
number is proxy itself)

Therefore, the port number in the "Invite" request-line is always same =
with the port number in the lower layer UDP. Why we insert redundant =
information? How about we drop this redundant and confusing port number =
in the URL? Please correct me if I am wrong. Thanks.

Best regards
Yuantao Zhang

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
Sent: 2008=D2=B46=EA=C56=EC=ED 19:55
To: Yuantao Zhang
Cc: Attila Sipos; sip@ietf.org; discussion@sipforum.org
Subject: Re: [Sip] What is the port number in "Invite" request-line? =
Thanks

The URI typically exists before the message is constructed. Then a
message is constructed and the URI included in it. The URI specifies
where the message is *intended* to go. And it might not go there
immediately. Instead it may first go to a proxy. When it gets there the
proxy needs to consult the R-URI to know where the message is intended
to go.

When the last hop to the final destination is made it is true that the
port number, and the ip address, may be redundant. But we don't expect
the intermediate nodes to remove the information. The recipient may do
as it wishes if it receives a request with addressing that is
inconsistent with where it is received.

	Paul

Yuantao Zhang wrote:
> Dear all
>=20
> Thanks for your reply. Please let me ask my question in another way.
>=20
> =
=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=3D=3D=3D=3D=3D=3D
>=20
> UDP:
>=20
> Src port: 5060
>=20
> Dst port: 5555        *=A9=AC------------
>=20
> =
=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=3D=3D=3D=3D=3D=3D
>=20
> SIP:
>=20
> INVITE sip:123101@192.168.100.100:5555 SIP/2.0  *=A9=AC------------
>=20
> Via: SIP/2.0/UDP 192.168.100.3:5060;rport;branch=3Dz9hG4bK27195
>=20
> From: <sip:123102@192.168.100.100>;tag=3D27266
>=20
> To: <sip:123101@192.168.100.100:5555>
>=20
> Call-ID: 24658@192.168.100.3
>=20
> CSeq: 20 INVITE
>=20
> Contact: <sip:123102@192.168.100.3:5060>
>=20
> =
=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=3D=3D=3D=3D=3D=3D
>=20
> The above "Invite" message is over UDP. In the UDP part (Transport=20
> layer), the Destination port is 5555. In the SIP part (application=20
> layer), the port in the "Invite" request-line URL is 5555 too. Both of =

> them are same. In which case the port in "Invite" request-line URL is=20
> different from the Destination port in lower layer UDP? Furthermore, =
if=20
> they are always same why we include redundant information (the port=20
> number) in "Invite" request-line URL? What happens if the port in=20
> "Invite" request-line URL is different from the Destination port in=20
> lower layer UDP?
>=20
> Thank you very much
>=20
> Best Regards
>=20
> Yuantao
>=20

------_=_NextPart_001_01C8C9D9.61BCB6BC
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charset=3Dks_c_5601-1987">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
6.5.7652.24">
<TITLE>RE: [Sip] What is the port number in &quot;Invite&quot; =
request-line? Thanks</TITLE>
</HEAD>
<BODY>
<!-- Converted from text/rtf format -->

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">------------------&nbsp;&nbsp;&nbsp;&nbsp; =
-------------------&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
-------------------</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">|UA1&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp; |SIP =
proxy&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; |&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |UA2&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &nbsp; |</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">|IP:192.168.100.1|-----|IP:</FONT> <FONT SIZE=3D1 FACE=3D"Courier =
New">192.168.100.2 |--------|IP: 192.168.100.3 |</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">|Port: 1111&nbsp;&nbsp;&nbsp;&nbsp; &nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp; |Port: =
2222&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |Port: =
3333&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">------------------&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
--------------------&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
-------------------</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier New">If =
UA1 makes call to UA2, UA1 should send the following message to the SIP =
prox</FONT><FONT SIZE=3D1 FACE=3D"Courier New">y:</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"fr"><FONT SIZE=3D1 FACE=3D"Courier =
New">=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=3D=3D=3D=3D=3D=3D</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"fr"><FONT SIZE=3D1 FACE=3D"Courier =
New">IP: </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"fr"><FONT SIZE=3D1 FACE=3D"Courier =
New">Src:</FONT><FONT SIZE=3D1 FACE=3D"Courier New"></FONT> <FONT =
SIZE=3D1 FACE=3D"Courier New">192.168.100.1</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"fr"><FONT SIZE=3D1 FACE=3D"Courier New">Dst: =
192.168.100.2</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"fr"><FONT SIZE=3D1 FACE=3D"Courier =
New">=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=3D=3D=3D=3D=3D=3D</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"fr"><FONT SIZE=3D1 FACE=3D"Courier =
New">UDP:</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"fr"><FONT SIZE=3D1 FACE=3D"Courier New">Src =
port:</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"fr"> <FONT SIZE=3D1 FACE=3D"Courier =
New">1111</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"fr"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"fr"><FONT SIZE=3D1 FACE=3D"Courier New">Dst =
port:</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"fr"> <FONT SIZE=3D1 FACE=3D"Courier =
New">2222</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"fr"><FONT SIZE=3D1 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN LANG=3D"fr"> =
&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D1 =
FACE=3D"Courier New">&nbsp;&nbsp; !</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"fr"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT =
SIZE=3D1 FACE=3D"Courier =
New">=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</FONT><FONT SIZE=3D1 FACE=3D"Courier =
New">=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</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">SIP:</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">INVITE sip:</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">UA2</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier New">@</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D1 =
FACE=3D"Courier New">192.168.100.2</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D1 =
FACE=3D"Courier New">:</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">2222</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier New"> =
SIP/2.0&nbsp;</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"> &nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D1 =
FACE=3D"Courier New">&nbsp;&nbsp; !</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">Via: SIP/2.0/UDP 192.168.100.</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D1 =
FACE=3D"Courier New">1</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier New">:</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D1 =
FACE=3D"Courier New">1111</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier New">;</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT =
SIZE=3D1 FACE=3D"Courier =
New">=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=3D=3D=3D=3D=3D=3D</FONT></SPAN></P>
<BR>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">Then SIP proxy sends the following message to =
UA2:</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"fr"><FONT SIZE=3D1 FACE=3D"Courier =
New">=3D=3D=3D=3D=3D=3D=3D=3D=3D</FONT><FONT SIZE=3D1 FACE=3D"Courier =
New">=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</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"fr"><FONT SIZE=3D1 FACE=3D"Courier =
New">IP: </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"fr"><FONT SIZE=3D1 FACE=3D"Courier =
New">Src:</FONT><FONT SIZE=3D1 FACE=3D"Courier New"></FONT> <FONT =
SIZE=3D1 FACE=3D"Courier New">192.168.100.2</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"fr"><FONT SIZE=3D1 FACE=3D"Courier New">Dst: =
192.168.100.3</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"fr"><FONT SIZE=3D1 FACE=3D"Courier =
New">=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=3D=3D=3D=3D=3D=3D</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"fr"><FONT SIZE=3D1 FACE=3D"Courier =
New">UDP:</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"fr"><FONT SIZE=3D1 FACE=3D"Courier New">Src =
port:</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"fr"> <FONT SIZE=3D1 FACE=3D"Courier =
New">2222</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"fr"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"fr"><FONT SIZE=3D1 FACE=3D"Courier New">Dst =
port:</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"fr"> <FONT SIZE=3D1 FACE=3D"Courier =
New">3333</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"fr"><FONT SIZE=3D1 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN LANG=3D"fr"> =
&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D1 =
FACE=3D"Courier New">&nbsp;&nbsp; !</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"fr"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT =
SIZE=3D1 FACE=3D"Courier =
New">=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=3D=3D=3D=3D=3D=3D</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">SIP:</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">INVITE sip:</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">UA2</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier New">@</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D1 =
FACE=3D"Courier New">192.168.100.2</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D1 =
FACE=3D"Courier New">:</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">3333</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier New"> =
SIP/2.0&nbsp;</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"> &nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D1 =
FACE=3D"Courier New">&nbsp;&nbsp; !</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">Via: SIP/2.0/UDP 192.168.100.</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D1 =
FACE=3D"Courier New">2</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier New">:</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D1 =
FACE=3D"Courier New">2222</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier New">;</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT =
SIZE=3D1 FACE=3D"Courier New">Via: SIP/2.0/UDP =
192.168.100.</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier New">1</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D1 =
FACE=3D"Courier New">:</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">1111</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier New">;</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT =
SIZE=3D1 FACE=3D"Courier =
New">=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=3D=3D=3D=3D=3D</FONT><FONT SIZE=3D1 =
FACE=3D"Courier New">=3D</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">UAC(UA1) does not know the</FONT> <FONT SIZE=3D1 FACE=3D"Courier =
New">port of</FONT> <FONT SIZE=3D1 FACE=3D"Courier New">UAS(UA2). All =
the UAC needs to know is the proxy's port. And the proxy knows all the =
ports of UAs under it (UAs tell proxy when register). Therefore, the =
proxy does not care the port number in th</FONT><FONT SIZE=3D1 =
FACE=3D"Courier New">e received</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"> <FONT SIZE=3D1 =
FACE=3D"Courier New">&quot;Invite&quot; request-line</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D1 =
FACE=3D"Courier New">n (because the port number is proxy =
itself)</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT =
SIZE=3D1 FACE=3D"Courier New">Therefore, the port number in =
the</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"> <FONT =
SIZE=3D1 FACE=3D"Courier New">&quot;Invite&quot; =
request-line</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier New"> is always same with =
the port number in the lower layer UDP. Why we insert redundant =
information?</FONT> <FONT SIZE=3D1 FACE=3D"Courier New">How about we =
drop this redundant and confusing port number</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D1 =
FACE=3D"Courier New"></FONT> <FONT SIZE=3D1 FACE=3D"Courier =
New">in</FONT><FONT SIZE=3D1 FACE=3D"Courier New"> the =
URL</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT =
SIZE=3D1 FACE=3D"Courier New">? Please correct me if I am wrong. =
Thanks.</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">Best regards</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">Yuantao Zhang</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT =
SIZE=3D1 FACE=3D"Courier New">-----Original Message-</FONT><FONT =
SIZE=3D1 FACE=3D"Courier New">----<BR>
From: Paul Kyzivat [<A =
HREF=3D"mailto:pkyzivat@cisco.com">mailto:pkyzivat@cisco.com</A>]<BR>
Sent: 2008</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D1 =
FACE=3D"=E1=E4&#20307;">=D2=B4</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D1 =
FACE=3D"Courier New">6</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D1 =
FACE=3D"=E1=E4&#20307;">=EA=C5</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D1 =
FACE=3D"Courier New">6</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D1 =
FACE=3D"=E1=E4&#20307;">=EC=ED</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D1 =
FACE=3D"Courier New"> 19:55<BR>
To:</FONT> <FONT SIZE=3D1 FACE=3D"Courier New">Yuantao Zhang<BR>
Cc: Attila Sipos; sip@ietf.org; discussion@sipforum.org<BR>
Subject: Re: [Sip] What is the port number in &quot;Invite&quot; =
request-line? Thanks</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">The URI typically ex</FONT><FONT SIZE=3D1 FACE=3D"Courier New">ists =
before the message is constructed. Then a</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">message is constructed and the URI included in it. The URI =
specifies</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">where the message is *intended* to go. And it might not go =
there</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">immediately. Instead it may first go to a proxy. When it gets there =
the</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">pro</FONT><FONT SIZE=3D1 FACE=3D"Courier New">xy needs to consult =
the R-URI to know where the message is intended</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier New">to =
go.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">When the last hop to the final destination is made it is true that =
the</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">port number, and the ip address, may be redundant. But we don't =
expect</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">the intermediate nodes to remove the i</FONT><FONT SIZE=3D1 =
FACE=3D"Courier New">nformation. The recipient may do</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier New">as =
it wishes if it receives a request with addressing that =
is</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">inconsistent with where it is received.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN =
LANG=3D"en-us">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D1 =
FACE=3D"Courier New">Paul</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">Yuantao Zhang wrote:</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; Dear all</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; T</FONT><FONT SIZE=3D1 FACE=3D"Courier New">hanks for your =
reply. Please let me ask my question in another way.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"fr"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"fr"><FONT SIZE=3D1 FACE=3D"Courier New">&gt; =
=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=3D=3D=3D=3D=3D=3D</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"fr"><FONT SIZE=3D1 FACE=3D"Courier New">&gt; =
</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"fr"><FONT SIZE=3D1 FACE=3D"Courier New">&gt; =
UDP:</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"fr"><FONT SIZE=3D1 FACE=3D"Courier New">&gt; =
</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"fr"><FONT SIZE=3D1 FACE=3D"Courier New">&gt; =
Src port: 5060</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"fr"><FONT SIZE=3D1 FACE=3D"Courier New">&gt; =
</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"fr"><FONT SIZE=3D1 FACE=3D"Courier New">&gt; =
Dst port: 5555&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
*=A9=AC</FONT><FONT SIZE=3D1 FACE=3D"Courier =
New">------------</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT =
SIZE=3D1 FACE=3D"Courier New">&gt; </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; =
=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</FONT><FONT SIZE=3D1 =
FACE=3D"Courier =
New">=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D</FON=
T></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; SIP:</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; INVITE sip:123101@192.168.100.100:5555 SIP/2.0&nbsp; =
*=A9=AC</FONT><FONT SIZE=3D1 FACE=3D"Courier =
New">------------</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; Via: SIP/2.0/UDP =
192.168.100.3:5060;rport;branch=3Dz9hG4bK27195</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; From: =
&lt;sip:123102@192.168.100.100&gt;;tag=3D27266</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; To: &lt;sip:123101@192.168.100.100:55</FONT><FONT SIZE=3D1 =
FACE=3D"Courier New">55&gt;</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; Call-ID: 24658@192.168.100.3</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; CSeq: 20 INVITE</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; Contact: =
&lt;sip:123102@192.168.100.3:5060&gt;</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; =
=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=3D=3D=3D=3D=3D=3D</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; The above &quot;Invite&quot; message is over UDP. In the UDP =
part (Transport </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; layer), the Destination port is 5555. In the SIP part =
(application </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; layer), the port in the &quot;Invite&quot; request-line URL is =
5555 too. Both of </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; them are same. In which case the port in &quot;Invite&quot; =
request-line URL is </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; different from the Destination port i</FONT><FONT SIZE=3D1 =
FACE=3D"Courier New">n lower layer UDP? Furthermore, if =
</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; they are always same why we include redundant information (the =
port </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; number) in &quot;Invite&quot; request-line URL? What happens =
if the port in </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; &quot;Invite&quot; request-line URL is different from the =
Destination port in </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; lower l</FONT><FONT SIZE=3D1 FACE=3D"Courier New">ayer =
UDP?</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; Thank you very much</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; Best Regards</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; Yuantao</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D1 FACE=3D"Courier =
New">&gt; </FONT></SPAN></P>

</BODY>
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_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
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From c.isopp@morgenbau.net  Mon Jun  9 04:35:46 2008
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Message-ID: <dotwgXHLPGXshelley.osborn@ietf.org>
From: "Rich Frederick" <shelley.osborn@ietf.org>
Reply-To: "Rich Frederick" <shelley.osborn@ietf.org>
To: shelley.osborn@ietf.org
Subject: April promo on watches
Date: Mon, 09 Jun 2008 07:35:55 -0500
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Date: Mon, 09 Jun 2008 07:52:43 -0400
From: Paul Kyzivat <pkyzivat@cisco.com>
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To: Yuantao Zhang <yuantao.zhang@ericsson.com>
References: <680808427CF931459462C3D78CB5EC6002461AEB@EXVS02.nasstar-t1.net>
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Subject: Re: [Sip] What is the port number in "Invite" request-line? Thanks
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Keith, Cullen, Scott, Vijay and I talked over the status of the domain- 
certs and EKU drafts this morning. Following teh conversation, we've  
agreed that as domain-certs  is normatively updating RFC 326i, it will  
be a standards-track document. We've amended the charter milestones to  
reflect this:

Change OLD
    Jan 2008  X.509 Certificates for TLS use in SIP to IESG (Info)
to NEW
    Sep 2008     X.509 Certificates for TLS use in SIP to IESG (PS)

We've also revised the milestone date for the companion EKU document:

Change date of:   X.509 extended key usage for SIP to IESG (PS)
to be Sep 2008

Thanks.

--
Dean

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Thread-Topic: [Sip] What is the port number in "Invite" request-line? Thanks
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From: "Yuantao Zhang" <yuantao.zhang@ericsson.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
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Subject: Re: [Sip] What is the port number in "Invite" request-line? Thanks
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On Tue, June 10, 2008 3:19 am, Yuantao Zhang wrote:
> Dear Paul
>
> Thanks for keeping answering my question. And it is happy to discuss with
> you. Could you give me a scenario that the port number in R-line and the
> lower layer UDP destination port are different? They are always same,
> aren't they? Thanks

No.  Look up 'Loose Routing' in RFC3261 (section 12.2.1.1 may help you). 
A request may be sent to destination other than the one indicated by the
Request-URI.  This may or may not involve different port numbers.

Regards,

Michael


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su, 2008-06-08 kello 22:08 +0200, ext cco kirjoitti:
> now, the call flows given as an example would function when the
> challenge (nonce) does not explicitly depend on the user which is
> actually trying to register itself, being generated
> (periodically) by the sip proxy itself; that is they are o.k. for an
> HTTP digest MD5 auth. as spec. in rfc 3261/rfc 2617. but they do NOT seem to 
> function in the real world when AKA is used, because:
> 
> 1. the challenge is user specific.
> 2. the "network" authenticates itself to the UE as well in case of AKA.

In an environment with plain RFC3310, an extra round trip would be
required. The main user of this spec, namely 3GPP fixes this by
requiring the presence of the Authorization field in the initial
REGISTER. 

Cheers,
Aki

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Hi,

=20

Can anybody help me which RFCs / TSs specifies SIP - INAP conversion.

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<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Can
anybody help me which RFCs / TSs specifies SIP &#8211; INAP =
conversion.<o:p></o:p></span></p>

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

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

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

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

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

<p class=3DMsoNormal><span =
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o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>NewNet
Communication Technologies, LLC<br>
<br>
Phone/VoIP: +41 43 544 07 19<br>
Fax: +41 41 790 40 39<br>
e-mail: <a href=3D"mailto:Paul.Heman@newnet.com"><span =
style=3D'color:blue'>Lukas.Alig@newnet.com</span></a><br>
</span><br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Office:
Floraweg 5<br>
CH-6343 Buonas<o:p></o:p></span></p>

<p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Switzerland<b=
r>
<br>
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to Recipient:&nbsp; This e-mail is confidential and meant only for the =
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<p class=3DMsoNormal><span lang=3DEN-US =
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may be a communication privileged by law.&nbsp; <o:p></o:p></span></p>

<p class=3DMsoNormal><span lang=3DEN-US =
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If you are not the intended recipient, please contact the sender by =
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<p class=3DMsoNormal><span lang=3DEN-US =
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<p class=3DMsoNormal><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p>

</div>

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_______________________________________________
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--===============1008524153==--


From sip-bounces@ietf.org  Tue Jun 10 03:27:07 2008
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From: cco <cristian.constantin@iptel.org>
To: Aki Niemi <aki.niemi@nokia.com>
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Cc: sip@ietf.org, Dragos Vingarzan <Dragos.Vingarzan@fokus.fraunhofer.de>
Subject: Re: [Sip] aka authentication and sip (rfc 3310)
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On Tue, Jun 10, 2008 at 11:55:43AM +0300, Aki Niemi wrote:
> su, 2008-06-08 kello 22:08 +0200, ext cco kirjoitti:
> > now, the call flows given as an example would function when the
> > challenge (nonce) does not explicitly depend on the user which is
> > actually trying to register itself, being generated
> > (periodically) by the sip proxy itself; that is they are o.k. for an
> > HTTP digest MD5 auth. as spec. in rfc 3261/rfc 2617. but they do NOT seem to 
> > function in the real world when AKA is used, because:
> > 
> > 1. the challenge is user specific.
> > 2. the "network" authenticates itself to the UE as well in case of AKA.
> 
> In an environment with plain RFC3310, an extra round trip would be
> required. The main user of this spec, namely 3GPP fixes this by

cristian: hi. aki, thanks a lot for the reply. I still have some
questions though. lets take it one step at a time. "an extra round trip
would be required" means that the call flow given as an example in the
rfc lacks that round trip, right? why?

> requiring the presence of the Authorization field in the initial
> REGISTER. 

cristian: why not specifying this in the rfc as well?

now, as I see it, the problem is that a lot of ue sip stacks implementors 
would take the rfc word for word and implement their software exactly 
how it is specified in the rfc. and in the end, everyone will wonder 
why the call flow in rfc3310 does not work or when it works it looks 
quite differently...

bye now!
cristian
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From sip-bounces@ietf.org  Tue Jun 10 05:05:40 2008
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Hi,
 
I've submitted a new version (-01) of the keep draft.
 
I can also be found at:
 
http://users.piuha.net/cholmber/drafts/draft-holmberg-sip-keep-01.txt
 
The major differences from the previous version are:
 
- I added text (chapter 8), provided by Vijay, which describes the relationship with connection reuse
 
- I added a Note (chapter 5), based on the comments from Juha, about usage of CRLF keep-alives even if there are no explicit indication.
 
Regards,
 
Christer
 
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Date: Tue, 10 Jun 2008 19:06:09 +0530
From: "Rammohan R" <rammohan.ravindranath@gmail.com>
To: "Lukas Alig" <lukas.alig@newnet.com>
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Cc: sip@ietf.org
Subject: Re: [Sip] INAP - SIP conversion
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Hi,

You can look at http://tools.ietf.org/html/rfc3976

Another paper that might be of help is
http://www.cs.columbia.edu/~hgs/papers/cucs-002-99.pdf

- Ram

On Tue, Jun 10, 2008 at 3:30 PM, Lukas Alig <lukas.alig@newnet.com> wrote:
> Hi,
>
>
>
> Can anybody help me which RFCs / TSs specifies SIP =96 INAP conversion.
>
>
>
> Thanks
>
>
>
> Lukas Alig
>
> Technical Consultant
>
>
>
> NewNet Communication Technologies, LLC
>
> Phone/VoIP: +41 43 544 07 19
> Fax: +41 41 790 40 39
> e-mail: Lukas.Alig@newnet.com
>
> Office: Floraweg 5
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>
_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From: atul garg <shivalk_80@yahoo.co.in>
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Subject: [Sip] Maximum allowed value of Clock rate in SDP ???
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Hello,

Is there any limitation about the maximum allowed value for the Clock-rate appears in SDP.

a = rtpmap : <payload type> <encoding name> </clock rate>

eg:- a =rtpmap:96 H263/90000

here this 90000, I have seen in real network ..is this value aloowed ???

If anyone also can mention any reference which states about this ,would help me alot.

Regards
shiva


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From: "Hul spencer" <hulbut@gmail.com>
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Subject: [Sip] Doubt in outbound draft v14 - auth proxy behaviour
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Hi,

I have a doubt related to outbound draft v14

In section 7 bullet 3

" If the proxy receives a final response from a branch other than a
      408 (Request Timeout) or a 430 (Flow Failed) response, the proxy
      MUST NOT forward the same request to another target representing
      the same AOR and instance-id.  The targeted instance has already
      provided its response."

As this section talks about authoritative proxy, this would imply that the
request gets sent to one of the edge proxy.
There can be cases where for same AOR and instance-id, UE register via 2
edge proxies, with reg-id=1 and reg-id=2.

  |----EP1---------|
  |                |
 UE        REGISTRAR/AUTH Proxy
  |                |
  |----EP2---------|

When auth proxy receives a request for this UE, it can decide to send the
request via EP1. Due to some problem(reboot etc) EP1, sends error response
404 Not Found(or any other error resp except 408 and 430). Once
authoritative proxy gets 404 , because of bullet 3 it can not try the other
path(via EP2) to UE.  Is it a correct behaiour ?  because  if authoritative
proxy sends  request to UE via EP2, request may still reach UE.


I understand that a proxy is supposed to send 430 in case flow is not there,
but there can be many other generic error conditions because of which proxy
may send any 4xx response.

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<span style="font-family: courier new,monospace;">Hi,</span><br style="font-family: courier new,monospace;"><br style="font-family: courier new,monospace;"><span style="font-family: courier new,monospace;">I have a doubt related to outbound draft v14</span><br style="font-family: courier new,monospace;">
<br style="font-family: courier new,monospace;"><span style="font-family: courier new,monospace;">In section 7 bullet 3 </span><br style="font-family: georgia,serif;"><pre style="font-family: arial,sans-serif;"><font size="1">&quot; If the proxy receives a final response from a branch other than a<br>
      408 (Request Timeout) or a 430 (Flow Failed) response, the proxy<br>      MUST NOT forward the same request to another target representing<br>      the same AOR and instance-id.  The targeted instance has already<br>
      provided its response.&quot;</font></pre><span style="font-family: courier new,monospace;">As this section talks about authoritative proxy, this would imply that the request gets sent to one of the edge proxy. </span><br style="font-family: courier new,monospace;">
<span style="font-family: courier new,monospace;">There can be cases where for same AOR and instance-id, UE register via 2 edge proxies, with reg-id=1 and reg-id=2.</span><br style="font-family: courier new,monospace;"><br style="font-family: courier new,monospace;">
<span style="font-family: courier new,monospace;">&nbsp; |----EP1---------|</span><br style="font-family: courier new,monospace;"><span style="font-family: courier new,monospace;">&nbsp; | &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &nbsp; &nbsp;&nbsp; |</span><br style="font-family: courier new,monospace;">
<span style="font-family: courier new,monospace;">&nbsp;UE&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; REGISTRAR/AUTH Proxy</span><br style="font-family: courier new,monospace;"><span style="font-family: courier new,monospace;">&nbsp; | &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &nbsp;&nbsp; |</span><br style="font-family: courier new,monospace;">
<span style="font-family: courier new,monospace;">&nbsp; |----EP2---------|</span><br style="font-family: courier new,monospace;"><span style="font-family: courier new,monospace;">&nbsp; </span><br style="font-family: courier new,monospace;">
<span style="font-family: courier new,monospace;">When auth proxy receives a request for this UE, it can decide to send the request via EP1. Due to some problem(reboot etc) EP1, sends error response 404 Not Found(or any other error resp except 408 and 430). Once authoritative proxy gets 404 , because of bullet 3 it can not try the other path(via EP2) to UE.&nbsp; Is it a correct behaiour ?&nbsp; because&nbsp; if authoritative proxy sends&nbsp; request to UE via EP2, request may still reach UE.</span><br>
<br><br style="font-family: courier new,monospace;"><span style="font-family: courier new,monospace;">I understand that a proxy is supposed to send 430 in case flow is not there, but there can be many other generic error conditions because of which proxy may send any 4xx response.</span><br>

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From: "Theo Zourzouvillys" <theo@crazygreek.co.uk>
To: "Yuantao Zhang" <yuantao.zhang@ericsson.com>
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Subject: Re: [Sip] What is the port number in "Invite" request-line? Thanks
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2008/6/10 Yuantao Zhang <yuantao.zhang@ericsson.com>:

> Thanks for keeping answering my question. And it is happy to discuss with you. Could you give me a scenario that the port number in R-line and the lower layer UDP destination port are different? They are always same, aren't they? Thanks

as well as others examples, there are also these times when a UAC
sends a request and the destination is resolved via SRV - the R-URI
may be sip:example.com, but the port could be anything at all.  Note
however that if a port is explicitly provided in the URI, then this
would not be the case.

 ~ Theo
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Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
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Christer,

I finally got around to this. A few comments:

Section 1:

    When SIP entities upstream
    receive the 2xx final response they will automatically terminate
    other associated outstanding early dialogs.

This isn't always wise. At this point it is still possible to receive 
additional 2xx responses. Discarding the early dialog is probably fine 
if subsequent 2xx responses with be dealt with by sending BYE. Retaining 
the early dialog isn't useful in that case. But if the UAC is planning 
to accept an additional 2xx response and retain the dialog rather than 
terminate it, then it should not discard the early dialog upon receipt 
of the first 2xx.

(Note that sending cancel upon receipt of a 2xx is MUST in 3261, but it 
is overridden by a 'Request-Disposition: no-cancel' specified in RFC 
3841. Terminating the early dialog at the UAC is also a MUST, but seems 
like it is probably an error.)

Section 4:

You give examples of kinds of resources that may be freed upon receipt 
of the 199. But you also hint at a problem:

    If the client is able to associate the 199 response with a specific
    media stream,

In many cases it may be hard/impossible to accomplish this. Can you give 
some plausible examples of cases when this will be possible?

I don't understand the following:

    The UAC MAY use additional information (for example the final
    response which triggered the 199 response) received in the 199
    response when initiating new sessions, if it is possible to avoid the
    new session initation request to be rejected.

I can't parse it into anything meaningful.

Section 5: When would it be advantageous for the UAS to send a 199?

Section 6:

This has MUST strength language. But of course not all proxies will 
implement this draft if/when it becomes an RFC. It it expected that this 
is to be an independent RFC to which support may be claimed?

Section 8:

    OPEN ISSUE: We need to discuss whether the To tag in the 199
    identifies the dialog that has been terminated, and/or if some
    additional information element (e.g.  SIP header) can be used, which
    would allow to indicate the termination of multiple dialogs in a
    single 199 response.

My preference is to use the To-tag, so that normal dialog matching rules 
will apply. We don't need more irregularities to complicate sip stacks.

    OPEN ISSUE: We need to discuss whether, in addition to the dialog
    identifier, some additional information (e.g. the non-2xx final
    response which triggered the 199 response) should be provided in the
    199 response.

I think it would be useful to at least acknowledge that a sipfrag can be 
carried in the 199, in a similar way to is carried in a refer event 
package notification. Would probably be good to state that it SHOULD NOT 
be included unless there was an Accept header mentioning the sipfrag 
content type in the request.

Section 9:

    A 199 Early Dialog Terminated provisional response MUST NOT contain a
    new SDP offer/answer message body, but the sender of the response MAY
    insert a copy of a previously sent offer/answer message body as
    otherwise allowed by the offer/answer rules for a provisional
    response.

    A forking proxy or a UAS MUST NOT trigger and send a 199 Early Dialog
    Terminated provisional response reliably in a situation where a
    reliable provisional response is required to contain an SDP offer/
    answer message body, according to [RFC3262] and [RFC3264].

These two paragraphs seem to be slightly at odds with one another, 
regarding the sending of an answer in a reliable 199. This can be fixed 
by changing the 2nd paragraph to only mention "offer", not "answer".

Section 10:

    OPEN ISSUE: We need to discuss whether a proxy should send 199
    reliably at all, since the proxy would have to terminate the
    associated PRACK, and support the logic associated with 100rel.  If
    not sent, how would required 100rel affect the usage of 199?

IMO it is a bad idea for a proxy to send a reliable 199. The problem is 
that you aren't sending the 199 unless there was already an early 
dialog. If there was, the proxy is not the second party in that dialog. 
The PRACK will be sent to the peer in the dialog, not the proxy. The 
proxy has no business terminating and responding to it. Yet the UAS 
won't be expecting it.

So I think it would only be possible to send a reliable 199 as part of 
some new dialog terminated by the proxy. But that then means the To-tag 
can't be used to indicate which dialog is being terminated. It just 
keeps getting more and more complicated.

And there really isn't any great value in sending this reliably. So 
don't do it.

	Thanks,
	Paul
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From: "Stuart Hoggan" <s.hoggan@CableLabs.com>
To: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>, <sip@ietf.org>
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All,

We have updated the document to incorporate feedback received so far.
It now contains a section that helps clarify the motivation.  Let us
know if you have further comments. 

-Stuart

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> DRAGE, Keith (Keith)
> Sent: Wednesday, June 04, 2008 3:48 AM
> To: sip@ietf.org
> Subject: Re: [Sip] draft-dotson-sip-mutual-auth-02
> 
> (As SIP WG cochair)
> 
> This has been raised in the SIP group a couple of times, and we have
not
> yet gained an idea of whether to proceed with it in the SIP WG or not.
> 
> There have been a couple of technical comments raised in the past from
> the security experts; my understanding is that these have now been
> clarified.
> 
> I believe in the past there has been an interested community that says
> this is useful. It has also been clarified that 3GPP would like to use
> it in their specifications.
> 
> So could I ask the WG to look at this document, and formally to
indicate
> whether they believe some work with this scope should be progressed as
a
> SIP WG item.
> 
>
http://www.ietf.org/internet-drafts/draft-dotson-sip-mutual-auth-02.txt
> 
> So if you support it please indicate to the SIP WG chairs.
> 
> If you have technical concerns (i.e. it should not proceed because it
is
> technically flawed beyond fixing) then please ideally mail the list,
> although the SIP WG chairs will also accept input.
> 
> Responses please in 1 calendar week from this date.
> 
> Regards
> 
> Keith
> 
> 
> ________________________________
> 
> 	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
> Behalf Of Stuart Hoggan
> 	Sent: Friday, May 02, 2008 5:15 PM
> 	To: sip@ietf.org
> 	Subject: [Sip] draft-dotson-sip-mutual-auth-02
> 
> 
> 
> 	Folks,
> 
> 
> 
> 	Based on the feedback received in Philadelphia, we have updated
> draft-dotson-sip-mutual-auth. The changes clarify the usage of the
> Proxy-Authentication-Info header, including support for multiple
values.
> 
> 
> 
> 
> 	Any further comments or suggestions will be appreciated. It
> would also be nice to obtain WG feedback on pursuing this I-D as a WG
> document.
> 
> 
> 
> 	Regards,
> 
> 	Stuart
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
_______________________________________________
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
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Hi Paul,
=20
>I finally got around to this. A few comments:
>
>Section 1:
>
>    When SIP entities upstream
>    receive the 2xx final response they will automatically terminate
>    other associated outstanding early dialogs.
>
>This isn't always wise. At this point it is still possible to receive
>additional 2xx responses. Discarding the early dialog is probably fine
>if subsequent 2xx responses with be dealt with by sending BYE.
=20
Correct. I can add some text, clarifying that the SIP enitty should
still be prepared to receive additional 2xx responses and deal with the
accordingly.
=20
=20
=20
>Retaining the early dialog isn't useful in that case. But if the UAC is
planning
>to accept an additional 2xx response and retain the dialog rather than
>terminate it, then it should not discard the early dialog upon receipt
>of the first 2xx.

I will mention that also.



>(Note that sending cancel upon receipt of a 2xx is MUST in 3261, but it
>is overridden by a 'Request-Disposition: no-cancel' specified in RFC
>3841. Terminating the early dialog at the UAC is also a MUST, but seems
>like it is probably an error.)

I can mention that RFC 3841 may affect the behavior of entities
regarding termination of early dialogs.


>Section 4:
>
>You give examples of kinds of resources that may be freed upon receipt
of the 199. But you also hint at a problem:
>
>   If the client is able to associate the 199 response with a specific
media stream,
>
>In many cases it may be hard/impossible to accomplish this. Can you
give some plausible examples of cases when this will be possible?

Maybe if the media stream is encrypted, and there are some encryption
parameters associated with the dialog. I am not an expert on that,
though....

Every now and then there have also been discussions of including
parameters, which are sent in the media stream, also in SIP messages, so
that media streams can be associated with an early dialog.

=20

>I don't understand the following:
>
>    The UAC MAY use additional information (for example the final
>    response which triggered the 199 response) received in the 199
>    response when initiating new sessions, if it is possible to avoid
the
>    new session initation request to be rejected.
>
>I can't parse it into anything meaningful.

This is related to the HERFP problem. I has been agreed to not solve
that problem in this draft, but it was agreed to add some words about
what the information could be used for.



>Section 5: When would it be advantageous for the UAS to send a 199?

This is one of the issues which was perviously discussed, so I listed
the possibility.

Francois also had that question, and I am ok with removing it.



>Section 6:
>
>This has MUST strength language. But of course not all proxies will
implement this draft if/when it becomes an RFC.

Well, the text only applies to proxies supporting the draft, so...

>It it expected that this is to be an independent RFC to which support
may be claimed?

Yes. The idea is NOT to make a normative reference to RFC3261.



>Section 8:
>
>    OPEN ISSUE: We need to discuss whether the To tag in the 199
>    identifies the dialog that has been terminated, and/or if some
>    additional information element (e.g.  SIP header) can be used,
which
>    would allow to indicate the termination of multiple dialogs in a
>    single 199 response.
>
>My preference is to use the To-tag, so that normal dialog matching
rules
>will apply. We don't need more irregularities to complicate sip stacks.

I agree. I am happy to remove the OPEN ISSUE, unless someone else has a
different opinion.


>    OPEN ISSUE: We need to discuss whether, in addition to the dialog
>    identifier, some additional information (e.g. the non-2xx final
>    response which triggered the 199 response) should be provided in
the
>    199 response.
>
>I think it would be useful to at least acknowledge that a sipfrag can
be
>carried in the 199, in a similar way to is carried in a refer event
>package notification. Would probably be good to state that it SHOULD
NOT
>be included unless there was an Accept header mentioning the sipfrag
>content type in the request.

This was discussed based on input from Francois just a few days ago, so
please take a look whether you are ok with the outcome of that
discussion.=20

We did not discuss the Accept header, though, and I think that is a good
point.



>Section 9:
>
>    A 199 Early Dialog Terminated provisional response MUST NOT contain
a
>    new SDP offer/answer message body, but the sender of the response
MAY
>    insert a copy of a previously sent offer/answer message body as
>    otherwise allowed by the offer/answer rules for a provisional
>   response.
>
>    A forking proxy or a UAS MUST NOT trigger and send a 199 Early
Dialog
>    Terminated provisional response reliably in a situation where a
>    reliable provisional response is required to contain an SDP offer/
>    answer message body, according to [RFC3262] and [RFC3264].
>
>These two paragraphs seem to be slightly at odds with one another,
>regarding the sending of an answer in a reliable 199. This can be fixed
>by changing the 2nd paragraph to only mention "offer", not "answer".

Ok



>Section 10:
>
>    OPEN ISSUE: We need to discuss whether a proxy should send 199
>    reliably at all, since the proxy would have to terminate the
>    associated PRACK, and support the logic associated with 100rel.  If
>    not sent, how would required 100rel affect the usage of 199?
>
>IMO it is a bad idea for a proxy to send a reliable 199. The problem is
>that you aren't sending the 199 unless there was already an early
>dialog. If there was, the proxy is not the second party in that dialog.
>The PRACK will be sent to the peer in the dialog, not the proxy. The
>proxy has no business terminating and responding to it. Yet the UAS
>won't be expecting it.
>
>So I think it would only be possible to send a reliable 199 as part of
>some new dialog terminated by the proxy. But that then means the To-tag
>can't be used to indicate which dialog is being terminated. It just
>keeps getting more and more complicated.
>
>And there really isn't any great value in sending this reliably. So
>don't do it.

I have no problem with that. A UA that doesn't support 199, but has
required 100rel, may discard the unreliable 199, but it really doesn't
matter.
=20
=20
Thank you very much for the comments!
=20
Regards,
=20
Christer
=20

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<DIV dir=3Dltr><FONT face=3DArial size=3D2>Hi Paul,</FONT></DIV>
<DIV dir=3Dltr><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV dir=3Dltr><FONT size=3D2>&gt;I finally got around to this. A few=20
comments:<BR>&gt;<BR>&gt;Section 1:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; =
When SIP=20
entities upstream<BR>&gt;&nbsp;&nbsp;&nbsp; receive the 2xx final =
response they=20
will automatically terminate<BR>&gt;&nbsp;&nbsp;&nbsp; other associated=20
outstanding early dialogs.<BR>&gt;<BR>&gt;This isn't always wise. At =
this point=20
it is still possible to receive<BR>&gt;additional 2xx responses. =
Discarding the=20
early dialog is probably fine<BR>&gt;if subsequent 2xx responses with be =
dealt=20
with by sending BYE.</FONT></DIV>
<DIV dir=3Dltr><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV dir=3Dltr><FONT size=3D2>Correct. I can add some text, clarifying =
that the SIP=20
enitty should still be prepared to receive additional 2xx responses and =
deal=20
with the accordingly.</FONT></DIV>
<DIV dir=3Dltr><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV dir=3Dltr><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV dir=3Dltr><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV dir=3Dltr><FONT size=3D2>&gt;Retaining the early dialog isn't =
useful in that=20
case. But if the UAC is planning<BR>&gt;to accept an additional 2xx =
response and=20
retain the dialog rather than<BR>&gt;terminate it, then it should not =
discard=20
the early dialog upon receipt<BR>&gt;of the first 2xx.</FONT></DIV><FONT =

size=3D2></FONT></DIV>
<P><FONT size=3D2>I will mention that also.<BR></FONT></P>
<P><FONT face=3DArial></FONT><BR>&gt;(Note that sending cancel upon =
receipt of a=20
2xx is MUST in 3261, but it<BR>&gt;is overridden by a =
'Request-Disposition:=20
no-cancel' specified in RFC<BR>&gt;3841. Terminating the early dialog at =
the UAC=20
is also a MUST, but seems<BR>&gt;like it is probably an error.)</P>
<P><FONT size=3D2>I can mention that RFC 3841 may affect the behavior of =
entities=20
regarding termination of early dialogs.</FONT></P>
<DIV><FONT face=3DArial size=3D2></FONT><FONT face=3DArial =
size=3D2></FONT><BR><FONT=20
size=3D2>&gt;Section 4:<BR>&gt;<BR>&gt;You give examples of kinds of =
resources=20
that may be freed upon receipt of the 199. But you also hint at a=20
problem:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp;If the client is able to =
associate the=20
199 response with a specific media stream,<BR>&gt;<BR>&gt;In many cases =
it may=20
be hard/impossible to accomplish this. Can you give some plausible =
examples of=20
cases when this will be possible?</FONT></DIV>
<P><FONT size=3D2>Maybe if the media stream is encrypted, and there are =
some=20
encryption parameters associated with the dialog. I am not an expert on =
that,=20
though....</FONT></P>
<P><FONT size=3D2>Every now and then there have also been discussions of =
including=20
parameters, which are sent in the media stream, also&nbsp;in SIP =
messages, so=20
that media streams can be associated with an early dialog.</FONT></P>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2></FONT><BR><FONT size=3D2>&gt;I don't =
understand the=20
following:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; The UAC MAY use additional=20
information (for example the final<BR>&gt;&nbsp;&nbsp;&nbsp; response =
which=20
triggered the 199 response) received in the =
199<BR>&gt;&nbsp;&nbsp;&nbsp;=20
response when initiating new sessions, if it is possible to avoid=20
the<BR>&gt;&nbsp;&nbsp;&nbsp; new session initation request to be=20
rejected.<BR>&gt;<BR>&gt;I can't parse it into anything =
meaningful.</FONT></DIV>
<P><FONT size=3D2>This is related to the HERFP problem. I has been =
agreed to not=20
solve that problem in this draft, but it was agreed to add some words =
about what=20
the information could be used for.</FONT></P>
<P><FONT size=3D2><BR><BR>&gt;Section 5: When would it be advantageous =
for the UAS=20
to send a 199?</FONT></P>
<P><SPAN class=3D293024905-11062008><FONT size=3D2>This is one of the =
issues which=20
was perviously discussed, so I listed the possibility.</FONT></SPAN></P>
<P><FONT size=3D2>Francois also had that question, and I am ok with =
removing=20
it.</FONT></P>
<P><FONT size=3D2><BR><BR>&gt;Section 6:<BR>&gt;<BR>&gt;This has MUST =
strength=20
language. But of course not all proxies will implement this draft =
if/when it=20
becomes an RFC.</FONT></P>
<P><FONT size=3D2>Well, the text only applies to proxies supporting the =
draft,=20
so...</FONT></P>
<P><FONT size=3D2>&gt;It it expected that this is to be an independent =
RFC to=20
which support may be claimed?</FONT></P>
<P><SPAN class=3D293024905-11062008></SPAN><FONT size=3D2>Y<SPAN=20
class=3D293024905-11062008>es. The idea is&nbsp;NOT to make a normative =
reference=20
to RFC3261.</SPAN></FONT></P>
<P><FONT size=3D2><SPAN =
class=3D293024905-11062008></SPAN><BR><BR>&gt;Section=20
8:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; OPEN ISSUE: We need to discuss =
whether the=20
To tag in the 199<BR>&gt;&nbsp;&nbsp;&nbsp; identifies the dialog that =
has been=20
terminated, and/or if some<BR>&gt;&nbsp;&nbsp;&nbsp; additional =
information=20
element (e.g.&nbsp; SIP header) can be used, =
which<BR>&gt;&nbsp;&nbsp;&nbsp;=20
would allow to indicate the termination of multiple dialogs in=20
a<BR>&gt;&nbsp;&nbsp;&nbsp; single 199 response.<BR>&gt;<BR>&gt;My =
preference is=20
to use the To-tag, so that normal dialog matching rules<BR>&gt;will =
apply. We=20
don't need more irregularities to complicate sip stacks.</FONT></P>
<P><FONT size=3D2>I agree. I am happy to remove the OPEN ISSUE, unless =
someone=20
else has a different opinion<SPAN =
class=3D293024905-11062008>.</SPAN></FONT></P>
<P><FONT face=3DArial size=3D2></FONT><BR><FONT =
size=3D2>&gt;&nbsp;&nbsp;&nbsp; OPEN=20
ISSUE: We need to discuss whether, in addition to the=20
dialog<BR>&gt;&nbsp;&nbsp;&nbsp; identifier, some additional information =
(e.g.=20
the non-2xx final<BR>&gt;&nbsp;&nbsp;&nbsp; response which triggered the =
199=20
response) should be provided in the<BR>&gt;&nbsp;&nbsp;&nbsp; 199=20
response.<BR>&gt;<BR>&gt;I think it would be useful to at least =
acknowledge that=20
a sipfrag can be<BR>&gt;carried in the 199, in a similar way to is =
carried in a=20
refer event<BR>&gt;package notification. Would probably be good to state =
that it=20
SHOULD NOT<BR>&gt;be included unless there was an Accept header =
mentioning the=20
sipfrag<BR>&gt;content type in the request.</FONT></P>
<P><FONT size=3D2>This was discussed based on input from Francois<SPAN=20
class=3D293024905-11062008> just a few days ago</SPAN>, so please take a =
look=20
whether you are ok with&nbsp;<SPAN class=3D293024905-11062008>the =
outcome of that=20
discussion. </SPAN></FONT></P>
<P><FONT size=3D2><SPAN class=3D293024905-11062008>We did not discuss =
the Accept=20
header, though, and I think that is a good point.</SPAN></FONT></P><FONT =

face=3DArial size=3D2></FONT>
<P><BR><BR><FONT size=3D2><SPAN =
class=3D293024905-11062008>&gt;</SPAN>Section=20
9:<BR><SPAN class=3D293024905-11062008></SPAN>&gt;<BR><SPAN=20
class=3D293024905-11062008>&gt;</SPAN>&nbsp;&nbsp;&nbsp; A 199 Early =
Dialog=20
Terminated provisional response MUST NOT contain a<BR><SPAN=20
class=3D293024905-11062008>&gt;</SPAN>&nbsp;&nbsp;&nbsp; new SDP =
offer/answer=20
message body, but the sender of the response MAY<BR><SPAN=20
class=3D293024905-11062008>&gt;</SPAN>&nbsp;&nbsp;&nbsp; insert a copy =
of a=20
previously sent offer/answer message body as<BR><SPAN=20
class=3D293024905-11062008>&gt;</SPAN>&nbsp;&nbsp;&nbsp; otherwise =
allowed by the=20
offer/answer rules for a provisional<BR><SPAN=20
class=3D293024905-11062008>&gt;</SPAN>&nbsp;&nbsp; response.<BR><SPAN=20
class=3D293024905-11062008></SPAN>&gt;<BR><SPAN=20
class=3D293024905-11062008>&gt;</SPAN>&nbsp;&nbsp;&nbsp; A forking proxy =
or a UAS=20
MUST NOT trigger and send a 199 Early Dialog<BR><SPAN=20
class=3D293024905-11062008>&gt;</SPAN>&nbsp;&nbsp;&nbsp; Terminated =
provisional=20
response reliably in a situation where a<BR><SPAN=20
class=3D293024905-11062008>&gt;</SPAN>&nbsp;&nbsp;&nbsp; reliable =
provisional=20
response is required to contain an SDP offer/<BR><SPAN=20
class=3D293024905-11062008>&gt;</SPAN>&nbsp;&nbsp;&nbsp; answer message =
body,=20
according to [RFC3262] and [RFC3264].<BR><SPAN=20
class=3D293024905-11062008></SPAN>&gt;<BR><SPAN=20
class=3D293024905-11062008>&gt;</SPAN>These two paragraphs seem to be =
slightly at=20
odds with one another,<BR><SPAN =
class=3D293024905-11062008>&gt;</SPAN>regarding=20
the sending of an answer in a reliable 199. This can be fixed<BR><SPAN=20
class=3D293024905-11062008>&gt;</SPAN>by changing the 2nd paragraph to =
only=20
mention "offer", not "answer".</FONT></P>
<P><SPAN class=3D293024905-11062008></SPAN><FONT size=3D2>O<SPAN=20
class=3D293024905-11062008>k</SPAN></FONT></P><SPAN=20
class=3D293024905-11062008></SPAN>
<P><BR><BR><FONT size=3D2><SPAN =
class=3D293024905-11062008>&gt;</SPAN>Section=20
10:<BR><SPAN class=3D293024905-11062008></SPAN>&gt;<BR><SPAN=20
class=3D293024905-11062008>&gt;</SPAN>&nbsp;&nbsp;&nbsp; OPEN ISSUE: We =
need to=20
discuss whether a proxy should send 199<BR><SPAN=20
class=3D293024905-11062008>&gt;</SPAN>&nbsp;&nbsp;&nbsp; reliably at =
all, since=20
the proxy would have to terminate the<BR><SPAN=20
class=3D293024905-11062008>&gt;</SPAN>&nbsp;&nbsp;&nbsp; associated =
PRACK, and=20
support the logic associated with 100rel.&nbsp; If<BR><SPAN=20
class=3D293024905-11062008>&gt;</SPAN>&nbsp;&nbsp;&nbsp; not sent, how =
would=20
required 100rel affect the usage of 199?<BR><SPAN=20
class=3D293024905-11062008></SPAN>&gt;<BR><SPAN=20
class=3D293024905-11062008>&gt;</SPAN>IMO it is a bad idea for a proxy =
to send a=20
reliable 199. The problem is<BR><SPAN =
class=3D293024905-11062008>&gt;</SPAN>that=20
you aren't sending the 199 unless there was already an early<BR><SPAN=20
class=3D293024905-11062008>&gt;</SPAN>dialog. If there was, the proxy is =
not the=20
second party in that dialog.<BR><SPAN =
class=3D293024905-11062008>&gt;</SPAN>The=20
PRACK will be sent to the peer in the dialog, not the proxy. =
The<BR><SPAN=20
class=3D293024905-11062008>&gt;</SPAN>proxy has no business terminating =
and=20
responding to it. Yet the UAS<BR><SPAN =
class=3D293024905-11062008>&gt;</SPAN>won't=20
be expecting it.<BR><SPAN =
class=3D293024905-11062008></SPAN>&gt;<BR><SPAN=20
class=3D293024905-11062008>&gt;</SPAN>So I think it would only be =
possible to send=20
a reliable 199 as part of<BR><SPAN =
class=3D293024905-11062008>&gt;</SPAN>some new=20
dialog terminated by the proxy. But that then means the To-tag<BR><SPAN=20
class=3D293024905-11062008>&gt;</SPAN>can't be used to indicate which =
dialog is=20
being terminated. It just<BR><SPAN =
class=3D293024905-11062008>&gt;</SPAN>keeps=20
getting more and more complicated.<BR><SPAN=20
class=3D293024905-11062008></SPAN>&gt;<BR><SPAN=20
class=3D293024905-11062008>&gt;</SPAN>And there really isn't any great =
value in=20
sending this reliably. So<BR><SPAN =
class=3D293024905-11062008>&gt;</SPAN>don't do=20
it.</FONT></P>
<DIV><SPAN class=3D293024905-11062008><FONT face=3DArial size=3D2>I have =
no problem=20
with that. A UA that doesn't support 199, but has required 100rel, may =
discard=20
the unreliable 199, but it really doesn't matte</FONT></SPAN><SPAN=20
class=3D293024905-11062008><FONT face=3DArial =
size=3D2>r.</FONT></SPAN></DIV>
<DIV><SPAN class=3D293024905-11062008><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D293024905-11062008><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D293024905-11062008><FONT face=3DArial size=3D2>Thank =
you very much=20
for the comments!</FONT></SPAN></DIV>
<DIV><SPAN class=3D293024905-11062008><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D293024905-11062008><FONT face=3DArial=20
size=3D2>Regards,</FONT></SPAN></DIV>
<DIV><SPAN class=3D293024905-11062008><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D293024905-11062008><FONT face=3DArial=20
size=3D2>Christer</FONT></SPAN></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV></NOSCRIPT>
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id=3Ddstb-id>if(typeof(dstb)!=3D "undefined"){ dstb();}</SCRIPT>
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Thread-Topic: [Sip] Draft submission: draft-ietf-sip-199-00
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From: "Attila Sipos" <attila.sipos@vegastream.com>
To: <sip@ietf.org>, "Christer Holmberg" <christer.holmberg@ericsson.com>
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( Aren't we at draft 04 now?  )
 
Should the document describe what should
happen if you get a 199 and the 199 refers
the only leg you have?


e.g. 
      UAC               P1                  UAS_2    UAS_3

       --- INVITE ------>
                         --- INVITE (leg 2) ->
                         --- INVITE (leg 3) ---------->
                         <-- 18x (leg 2) -----
       <-- 18x (leg 2) --
                         <-- 4xx (leg 2) -----
       <-- 199 (leg 2) --

 =======================
  what does UAC do now?
 =======================

                         <-- 18x (leg 3) --------------
       <-- 18x (leg 3) --

                         <-- 200 (leg 3) --------------
       <-- 200 (leg 3) --
       --- ACK (leg 3) ->
                         --- ACK (leg 3) ------------->


The UAC would have to act as if it hadn't received
the "18x (leg 2)" and "199 (leg 2)" message.  So it
would be as if it was still be waiting for its first
non-100 response.

This answer is obvious maybe but maybe some explicit
text might be useful otherwise the question might come
up over and over.

Regards,

Attila

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Hi,

BYE sipua"<sip:100.0.10.14:5070  SIP/2.0
Via: S IP/2.0/UDP 100.172 .1.109:5060;bra nch=z87uytG4bK-7887695638

.
.
.

here in the first line -> sipua"<<sip:100.0.10.14:5070  SIP/2.0 is a valid Request /Responseline ???

Regards
Shiva


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no.


-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
atul garg
Sent: 11 June 2008 10:56
To: sip@ietf.org
Subject: [Sip] Is this a valid Request Line/response lie ??

Hi,

BYE sipua"<sip:100.0.10.14:5070  SIP/2.0
Via: S IP/2.0/UDP 100.172 .1.109:5060;bra nch=z87uytG4bK-7887695638

.
.
.

here in the first line -> sipua"<<sip:100.0.10.14:5070  SIP/2.0 is a
valid Request /Responseline ???

Regards
Shiva


      Unlimited freedom, unlimited storage. Get it now, on
http://help.yahoo.com/l/in/yahoo/mail/yahoomail/tools/tools-08.html/
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sip-implementors@cs.columbia.edu for questions on current sip Use
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From sip-bounces@ietf.org  Wed Jun 11 07:08:55 2008
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From: Colin Perkins <csp@csperkins.org>
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Subject: Re: [Sip] Maximum allowed value of Clock rate in SDP ???
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On 10 Jun 2008, at 14:55, atul garg wrote:
> Is there any limitation about the maximum allowed value for the  
> Clock-rate appears in SDP.
>
> a = rtpmap : <payload type> <encoding name> </clock rate>
>
> eg:- a =rtpmap:96 H263/90000
>
> here this 90000, I have seen in real network ..is this value  
> aloowed ???

Yes - this is the clock rate used for most video formats.

> If anyone also can mention any reference which states about  
> this ,would help me alot.

The legal values for the clock rate for each RTP payload format are  
defined in the payload format RFC. SDP doesn't specify a maximum  
value. The largest clock rate used that I'm aware of is video/ 
smpte292m, which uses a rate of 148500000.

-- 
Colin Perkins
http://csperkins.org/



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Hi, 

>( Aren't we at draft 04 now?  )

No. We are now discussing the SIP WG draft, which has been put together
based on the requirement in the previous individual draft for SIPPING.

>Should the document describe what should happen if you get a 199 and
the 199 refers the only leg you have?
>
>
>e.g. 
>      UAC               P1                  UAS_2    UAS_3
>
>       --- INVITE ------>
>                         --- INVITE (leg 2) ->
>                         --- INVITE (leg 3) ---------->
>                         <-- 18x (leg 2) -----
>       <-- 18x (leg 2) --
>                         <-- 4xx (leg 2) -----
>       <-- 199 (leg 2) --
>
> =======================
>  what does UAC do now?
> =======================
>
>                         <-- 18x (leg 3) --------------
>       <-- 18x (leg 3) --
>
>                         <-- 200 (leg 3) --------------
>       <-- 200 (leg 3) --
>       --- ACK (leg 3) ->
>                         --- ACK (leg 3) ------------->
>
>
>The UAC would have to act as if it hadn't received the "18x (leg 2)"
and "199 (leg 2)" message.  So it would be as if it >was still be
waiting for its first non-100 response.
>
>This answer is obvious maybe but maybe some explicit text might be
useful otherwise the question might come up over and 
>over.

I can add some words about it.

Thanks for the comment!

Regards,

Christer

Regards,

Attila

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I have a basic SIP question.  The way I understood RFC3261, the "To:" header
field in a response  has to match the "To:" header field of the
corresponding request (if no To tag is present in the request).  This is
true for 99% of the SIP conversations I've seen but not for the (real-life)
example below (note that the "To:" header fields differ).  Wireshark has no
problem with analyzing this SIP conversation, so I suspect I'm
misinterpreting the RFC.

Can the "To:" header fields be different, and if so, how can the response be 
correlated to the request?


#
# The request
#
INVITE sip:5557451576@e.voncp.com:10000 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.156:10000;branch=z9hG4bK-762646f2
From: 555-482-0690
<sip:15554820690@e.voncp.com:10000>;tag=716ff48d4ef52332o0
To: <sip:5557451576@e.voncp.com:10000>
Remote-Party-ID: 555-482-0690
<sip:15554820690@e.voncp.com:10000>;screen=yes;party=calling
Call-ID: 81f537e9-aaa35786@192.168.5.156
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest
username="15554820690",realm="69.59.242.85",nonce="1489356920",uri="sip:5557451576@e.voncp.com:10000",algorithm=MD5,response="095bb689741268d970ca807964d58a73"
Contact: 555-482-0690 <sip:15554820690@192.168.5.156:10000>
Expires: 240
User-Agent: 001310757539 Linksys/RT31P2-3.1.6(LI)
Content-Length: 316


#
# The response
#
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.5.156:10000;branch=z9hG4bK-762646f2
From: 555-482-0690
<sip:15554820690@e.voncp.com:10000>;tag=716ff48d4ef52332o0;natted=24.235.117.242
To: <sip:0vonCallee-2351073@e.voncp.com:10000>;tag=1645248237
Call-ID: 81f537e9-aaa35786@192.168.5.156
CSeq: 102 INVITE
Contact: <sip:0vonCallee-2351094@69.59.242.85:10000>
Max-Forwards: 15



#
# The relevant RFC section
#
8.2.6.2 Headers and Tags


   The From field of the response MUST equal the From header field of
   the request.  The Call-ID header field of the response MUST equal the
   Call-ID header field of the request.  The CSeq header field of the
   response MUST equal the CSeq field of the request.  The Via header
   field values in the response MUST equal the Via header field values
   in the request and MUST maintain the same ordering.

   If a request contained a To tag in the request, the To header field
   in the response MUST equal that of the request.  However, if the To
   header field in the request did not contain a tag, the URI in the To
   header field in the response MUST equal the URI in the To header
   field; additionally, the UAS MUST add a tag to the To header field in
   the response (with the exception of the 100 (Trying) response, in
   which a tag MAY be present).  This serves to identify the UAS that is
   responding, possibly resulting in a component of a dialog ID.  The
   same tag MUST be used for all responses to that request, both final
   and provisional (again excepting the 100 (Trying)).  Procedures for
   the generation of tags are defined in Section 19.3. 

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


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Subject: Re: [Sip] Request/response question...
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barcaroller@sympatico.ca wrote:
> I have a basic SIP question.  The way I understood RFC3261, the "To:" header
> field in a response  has to match the "To:" header field of the
> corresponding request (if no To tag is present in the request).  This is
> true for 99% of the SIP conversations I've seen but not for the (real-life)
> example below (note that the "To:" header fields differ).  Wireshark has no
> problem with analyzing this SIP conversation, so I suspect I'm
> misinterpreting the RFC.
> 
> Can the "To:" header fields be different, and if so, how can the response be 
> correlated to the request?

The To in the response can only differ from that in the request because 
of the addition of a tag= parameter if there wasn't one in the request. 
The rest of the To-header must be the same.

The matching of the response to the request is based on the branch 
parameter in the Via, so that isn't the problem.

	Paul

> 
> #
> # The request
> #
> INVITE sip:5557451576@e.voncp.com:10000 SIP/2.0
> Via: SIP/2.0/UDP 192.168.5.156:10000;branch=z9hG4bK-762646f2
> From: 555-482-0690
> <sip:15554820690@e.voncp.com:10000>;tag=716ff48d4ef52332o0
> To: <sip:5557451576@e.voncp.com:10000>
> Remote-Party-ID: 555-482-0690
> <sip:15554820690@e.voncp.com:10000>;screen=yes;party=calling
> Call-ID: 81f537e9-aaa35786@192.168.5.156
> CSeq: 102 INVITE
> Max-Forwards: 70
> Proxy-Authorization: Digest
> username="15554820690",realm="69.59.242.85",nonce="1489356920",uri="sip:5557451576@e.voncp.com:10000",algorithm=MD5,response="095bb689741268d970ca807964d58a73"
> Contact: 555-482-0690 <sip:15554820690@192.168.5.156:10000>
> Expires: 240
> User-Agent: 001310757539 Linksys/RT31P2-3.1.6(LI)
> Content-Length: 316
> 
> 
> #
> # The response
> #
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.5.156:10000;branch=z9hG4bK-762646f2
> From: 555-482-0690
> <sip:15554820690@e.voncp.com:10000>;tag=716ff48d4ef52332o0;natted=24.235.117.242
> To: <sip:0vonCallee-2351073@e.voncp.com:10000>;tag=1645248237
> Call-ID: 81f537e9-aaa35786@192.168.5.156
> CSeq: 102 INVITE
> Contact: <sip:0vonCallee-2351094@69.59.242.85:10000>
> Max-Forwards: 15
> 
> 
> 
> #
> # The relevant RFC section
> #
> 8.2.6.2 Headers and Tags
> 
> 
>    The From field of the response MUST equal the From header field of
>    the request.  The Call-ID header field of the response MUST equal the
>    Call-ID header field of the request.  The CSeq header field of the
>    response MUST equal the CSeq field of the request.  The Via header
>    field values in the response MUST equal the Via header field values
>    in the request and MUST maintain the same ordering.
> 
>    If a request contained a To tag in the request, the To header field
>    in the response MUST equal that of the request.  However, if the To
>    header field in the request did not contain a tag, the URI in the To
>    header field in the response MUST equal the URI in the To header
>    field; additionally, the UAS MUST add a tag to the To header field in
>    the response (with the exception of the 100 (Trying) response, in
>    which a tag MAY be present).  This serves to identify the UAS that is
>    responding, possibly resulting in a component of a dialog ID.  The
>    same tag MUST be used for all responses to that request, both final
>    and provisional (again excepting the 100 (Trying)).  Procedures for
>    the generation of tags are defined in Section 19.3. 
> 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


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From: Dale.Worley@comcast.net
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Subject: Re: [Sip] Request/response question...
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   From: <barcaroller@sympatico.ca>

   I have a basic SIP question.  The way I understood RFC3261, the "To:" header
   field in a response  has to match the "To:" header field of the
   corresponding request (if no To tag is present in the request).  This is
   true for 99% of the SIP conversations I've seen but not for the (real-life)
   example below (note that the "To:" header fields differ).  Wireshark has no
   problem with analyzing this SIP conversation, so I suspect I'm
   misinterpreting the RFC.

   Can the "To:" header fields be different, and if so, how can the
   response be correlated to the request?

The RFC does require the To (and From) headers to match, but everybody
believes that is an obsolete requirement and will probably be relaxed
in the near future.  So generate requests with matching To/From
headers, but do not depend on them.

Requests and responses are correlated by the branch parameters in
their Via headers.

Requests are grouped into dialogs by their "dialog identifiers", that
is, call-id/to-tag/from-tag.

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


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Thread-Topic: [Sip] Request/response question...
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From: "Francois Audet" <audet@nortel.com>
To: <barcaroller@sympatico.ca>, <sip@ietf.org>
Subject: Re: [Sip] Request/response question...
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See section 3 of http://www.ietf.org/rfc/rfc4916.txt. 

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of barcaroller@sympatico.ca
> Sent: Wednesday, June 11, 2008 14:55
> To: sip@ietf.org
> Subject: [Sip] Request/response question...
> 
> 
> I have a basic SIP question.  The way I understood RFC3261, 
> the "To:" header field in a response  has to match the "To:" 
> header field of the corresponding request (if no To tag is 
> present in the request).  This is true for 99% of the SIP 
> conversations I've seen but not for the (real-life) example 
> below (note that the "To:" header fields differ).  Wireshark 
> has no problem with analyzing this SIP conversation, so I 
> suspect I'm misinterpreting the RFC.
> 
> Can the "To:" header fields be different, and if so, how can 
> the response be correlated to the request?
> 
> 
> #
> # The request
> #
> INVITE sip:5557451576@e.voncp.com:10000 SIP/2.0
> Via: SIP/2.0/UDP 192.168.5.156:10000;branch=z9hG4bK-762646f2
> From: 555-482-0690
> <sip:15554820690@e.voncp.com:10000>;tag=716ff48d4ef52332o0
> To: <sip:5557451576@e.voncp.com:10000>
> Remote-Party-ID: 555-482-0690
> <sip:15554820690@e.voncp.com:10000>;screen=yes;party=calling
> Call-ID: 81f537e9-aaa35786@192.168.5.156
> CSeq: 102 INVITE
> Max-Forwards: 70
> Proxy-Authorization: Digest
> username="15554820690",realm="69.59.242.85",nonce="1489356920"
> ,uri="sip:5557451576@e.voncp.com:10000",algorithm=MD5,response
="095bb689741268d970ca807964d58a73"
> Contact: 555-482-0690 <sip:15554820690@192.168.5.156:10000>
> Expires: 240
> User-Agent: 001310757539 Linksys/RT31P2-3.1.6(LI)
> Content-Length: 316
> 
> 
> #
> # The response
> #
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.5.156:10000;branch=z9hG4bK-762646f2
> From: 555-482-0690
> <sip:15554820690@e.voncp.com:10000>;tag=716ff48d4ef52332o0;nat
> ted=24.235.117.242
> To: <sip:0vonCallee-2351073@e.voncp.com:10000>;tag=1645248237
> Call-ID: 81f537e9-aaa35786@192.168.5.156
> CSeq: 102 INVITE
> Contact: <sip:0vonCallee-2351094@69.59.242.85:10000>
> Max-Forwards: 15
> 
> 
> 
> #
> # The relevant RFC section
> #
> 8.2.6.2 Headers and Tags
> 
> 
>    The From field of the response MUST equal the From header field of
>    the request.  The Call-ID header field of the response 
> MUST equal the
>    Call-ID header field of the request.  The CSeq header field of the
>    response MUST equal the CSeq field of the request.  The Via header
>    field values in the response MUST equal the Via header field values
>    in the request and MUST maintain the same ordering.
> 
>    If a request contained a To tag in the request, the To header field
>    in the response MUST equal that of the request.  However, if the To
>    header field in the request did not contain a tag, the URI 
> in the To
>    header field in the response MUST equal the URI in the To header
>    field; additionally, the UAS MUST add a tag to the To 
> header field in
>    the response (with the exception of the 100 (Trying) response, in
>    which a tag MAY be present).  This serves to identify the 
> UAS that is
>    responding, possibly resulting in a component of a dialog ID.  The
>    same tag MUST be used for all responses to that request, both final
>    and provisional (again excepting the 100 (Trying)).  Procedures for
>    the generation of tags are defined in Section 19.3. 
> 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
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Subject: Re: [Sip] Request/response question...
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Dale.Worley@comcast.net wrote:
>    From: <barcaroller@sympatico.ca>
> 
>    I have a basic SIP question.  The way I understood RFC3261, the "To:" header
>    field in a response  has to match the "To:" header field of the
>    corresponding request (if no To tag is present in the request).  This is
>    true for 99% of the SIP conversations I've seen but not for the (real-life)
>    example below (note that the "To:" header fields differ).  Wireshark has no
>    problem with analyzing this SIP conversation, so I suspect I'm
>    misinterpreting the RFC.
> 
>    Can the "To:" header fields be different, and if so, how can the
>    response be correlated to the request?
> 
> The RFC does require the To (and From) headers to match, but everybody
> believes that is an obsolete requirement and will probably be relaxed
> in the near future.  So generate requests with matching To/From
> headers, but do not depend on them.

AFAIK there is no agreement that they shouldn't match in a request and 
its *response*. Most do now agree that they might change for subsequent 
requests within a dialog.

At one point there may have been a little talk about changing the To in 
a response to indicate the actual identity of the called party. But once 
it was realized there was no good way to authenticate that we stopped 
talking about it. (At least that is what I remember.)

	Paul

> Requests and responses are correlated by the branch parameters in
> their Via headers.
> 
> Requests are grouped into dialogs by their "dialog identifiers", that
> is, call-id/to-tag/from-tag.
> 
> Dale
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
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On Jun 11, 2008, at 5:27 PM, Paul Kyzivat wrote:

>
>
> Dale.Worley@comcast.net wrote:
>>   From: <barcaroller@sympatico.ca>
>>
>>   I have a basic SIP question.  The way I understood RFC3261, the  
>> "To:" header
>>   field in a response  has to match the "To:" header field of the
>>   corresponding request (if no To tag is present in the request).   
>> This is
>>   true for 99% of the SIP conversations I've seen but not for the  
>> (real-life)
>>   example below (note that the "To:" header fields differ).   
>> Wireshark has no
>>   problem with analyzing this SIP conversation, so I suspect I'm
>>   misinterpreting the RFC.
>>
>>   Can the "To:" header fields be different, and if so, how can the
>>   response be correlated to the request?
>>
>> The RFC does require the To (and From) headers to match, but  
>> everybody
>> believes that is an obsolete requirement and will probably be relaxed
>> in the near future.  So generate requests with matching To/From
>> headers, but do not depend on them.
>
> AFAIK there is no agreement that they shouldn't match in a request and
> its *response*. Most do now agree that they might change for  
> subsequent
> requests within a dialog.
>
> At one point there may have been a little talk about changing the To  
> in
> a response to indicate the actual identity of the called party. But  
> once
> it was realized there was no good way to authenticate that we stopped
> talking about it. (At least that is what I remember.)
>

Changing it in a response is critical to extending RFC 4474 for use in  
responses where there are certificates available in the UAs or where  
identity-based signatures are used.

In other words, IF we solve the PKI problem and don't get distracted  
by the unanticipated respondent problem, then we NEED variant To in  
responses to be able to se response-identity, because connected- 
identity is a) late in a dialog and b) useless for non-dialog  
transactions, like MESSAGE.

--
Dean

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Subject: Re: [Sip] aka authentication and sip (rfc 3310)
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ti, 2008-06-10 kello 12:27 +0200, ext cco kirjoitti:
> On Tue, Jun 10, 2008 at 11:55:43AM +0300, Aki Niemi wrote:
> > su, 2008-06-08 kello 22:08 +0200, ext cco kirjoitti:
> > > now, the call flows given as an example would function when the
> > > challenge (nonce) does not explicitly depend on the user which is
> > > actually trying to register itself, being generated
> > > (periodically) by the sip proxy itself; that is they are o.k. for an
> > > HTTP digest MD5 auth. as spec. in rfc 3261/rfc 2617. but they do NOT seem to 
> > > function in the real world when AKA is used, because:
> > > 
> > > 1. the challenge is user specific.
> > > 2. the "network" authenticates itself to the UE as well in case of AKA.
> > 
> > In an environment with plain RFC3310, an extra round trip would be
> > required. The main user of this spec, namely 3GPP fixes this by
> 
> cristian: hi. aki, thanks a lot for the reply. I still have some
> questions though. lets take it one step at a time. "an extra round trip
> would be required" means that the call flow given as an example in the
> rfc lacks that round trip, right? why?

Well, the REGISTER header is not visible, so it indeed could contain an
Authorization field, as it normally would if there are cached
credentials.

> > requiring the presence of the Authorization field in the initial
> > REGISTER. 
> 
> cristian: why not specifying this in the rfc as well?

I don't think it matters. I can't see Digest AKA very useful beyond 3GPP
IMS anyway, which means implementing a bunch of their TSs.

> now, as I see it, the problem is that a lot of ue sip stacks implementors 
> would take the rfc word for word and implement their software exactly 
> how it is specified in the rfc. 

Not necessarily a bad thing...

> and in the end, everyone will wonder 
> why the call flow in rfc3310 does not work or when it works it looks 
> quite differently...

There isn't a big difference, really. Look at it as there always being a
set of cached credentials at the UA that are included in every REGISTER.
It just happens that the UA always gets a 401 in response.

Cheers,
Aki

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Hi Everybody.

I'm currently working on SIP/TDM Gateway application.
I've encountered following problem.
During Sip dialog both sides at the exactly same moment send Re-Invite 
requests.
Those are reaching their destinations while both client are expecting 
replies. What should be the behavior of SIP client in a situation like that.



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Hello,

I have a question regarding push to talk message .

Is there any way through which I can identify that particular SIP message is meant for PTT ie. this sip message is a part of some Push To Talk session..

Like through which I can filter that such message is for PTT.

Such criteria which can be applicable on all SIP messages ie.

1) Invite
2) Register
3) 2xx/1xx/4xx/3xx etc. responces
..
..
etc.

Thnx
Atul



      From Chandigarh to Chennai - find friends all over India. Go to http://in.promos.yahoo.com/groups/citygroups/

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Hello all,
I got a few questions as below want to hear your inputs:

Is there anyone implemented SIP to support long call audit? (e.g. SIP
INFO, SIP re-INVITE with SDP)?

How do you implement SIP to support the use of the DIVERSION header
and under what circumtances this DIVERSION header is been provaked?
(Such as call forwading?) If so, do you support
sending privacy information via the DIVERSION header?

Thanks for sharing.

Chris

On 6/4/08, Yuantao Zhang <yuantao.zhang@ericsson.com> wrote:
> Dear all
>
>
>
> INVITE sip:bob@biloxi.com SIP/2.0:5555
>
> INVITE sip:bob@biloxi.com SIP/2.0
>
>
>
> Above are 2 "Invite" request-line examples. In example one, it indicates
> a port number, 5555. What does the port number mean?
>
>
>
> Is it the port number of SIP proxy? Why we include port number here? The
> SIP proxy(biloxi.com) application layer who receives this "Invite"
> should receive this via port 5555 from lowe layer(UDP). Therefore, The
> SIP proxy(biloxi.com) application layer already know the port is 5555.
> Actualy, the SIP proxy application layer keeps listening port 5555.
>
>
>
> The second "Invite" request-line example is from RFC3261 and does not
> include port number. Can I say the port number is useless? So why
> example one includes a useless part in "Invite" request-line URL?
>
>
>
> Or is the port number is which the UAS is listening to? How the UAC
> knows it is 5555 if so?
>
>
>
> Any standard on this port number issue? Thanks.
>
>
>
> Best regards
>
> Steven
>
>

-- 
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Subject: Re: [Sip] [SIPForum-discussion] What is the port number in "Invite"
	request-line? Thanks
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Hey Yuantao,


>  *From:* discussion-bounces@sipforum.org [mailto:
> discussion-bounces@sipforum.org] *On Behalf Of *Yuantao Zhang
> *Sent:* Wednesday, June 04, 2008 12:10
> *To:* sip@ietf.org; discussion@sipforum.org
> *Subject:* [SIPForum-discussion] What is the port number in "Invite"
> request-line? Thanks
>
> Dear all
>
> INVITE sip:bob@biloxi.com <sip%3Abob@biloxi.com> SIP/2.0:5555
>
> INVITE sip:bob@biloxi.com <sip%3Abob@biloxi.com> SIP/2.0
>
> Above are 2 "Invite" request-line examples. In example one, it indicates a
> port number, 5555. What does the port number mean? Is it the port number
> of SIP proxy?
>
As you suspected yourself and as Shmulik pointed correctly, the number 555
indicates
the port number of the running SIP Proxy which the UA is registered to.

>  Why we include port number here? The SIP proxy(biloxi.com) application
> layer who receives
>
Simply because this is a non-standard SIP port, just as if a web server was
running on port
8081 you would have to explicitly tell the browser to use that port, for
example:
http://www.non-standard-server.com:8081

This is also required when more than one SIP Proxy is involved in the
process and the
INVITE travels from one proxy to another.

  The second "Invite" request-line example is from RFC3261 and does not
> include port number. Can I say the port number is useless? So why example
> one includes a useless part in "Invite" request-line URL?
>
To be more precise the first invite message (INVITE
sip:bob@biloxi.com<sip%3Abob@biloxi.com>SIP/2.0:5555)
is wrong. It should have been INVITE sip:bob@biloxi.com:5555 SIP/2.0

>  Any standard on this port number issue? Thanks.
>

It is very common for different SIP stacks to behave somewhat differently
and quite
a few of them do not comply with the RFC which is very unfortunate.




-- 
-- 
Regards,

*Liran Tal
**Unix Kernel Developer
NGN Solutions, TELES Israel

*

[image: [Teles Logo]]

Phone           +972 4 9092592
Fax               +972 4 9597144
Mobile           +972 52 3793446

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Hey Yuantao,<br><br><div class="gmail_quote"><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="red" lang="EN-US"><div><br>

<div>

<div style="border-style: solid none none; border-color: rgb(181, 196, 223) -moz-use-text-color -moz-use-text-color; border-width: 1pt medium medium; padding: 3pt 0in 0in;">

<p><b><span style="font-size: 10pt;">From:</span></b><span style="font-size: 10pt;">
<a href="mailto:discussion-bounces@sipforum.org" target="_blank">discussion-bounces@sipforum.org</a> [mailto:<a href="mailto:discussion-bounces@sipforum.org" target="_blank">discussion-bounces@sipforum.org</a>] <b>On
Behalf Of </b>Yuantao Zhang<br>
<b>Sent:</b> Wednesday, June 04, 2008 12:10<br>
<b>To:</b> <a href="mailto:sip@ietf.org" target="_blank">sip@ietf.org</a>; <a href="mailto:discussion@sipforum.org" target="_blank">discussion@sipforum.org</a><br>
<b>Subject:</b> [SIPForum-discussion] What is the port number in
&quot;Invite&quot; request-line? Thanks</span></p>

</div>

</div>



<p><span style="font-size: 10pt; color: navy;">Dear all</span></p>



<p><span style="font-size: 10pt; color: navy;">INVITE <a href="mailto:sip%3Abob@biloxi.com" target="_blank">sip:bob@biloxi.com</a> SIP/2.0:5555</span></p>

<p><span style="font-size: 10pt; color: navy;">INVITE <a href="mailto:sip%3Abob@biloxi.com" target="_blank">sip:bob@biloxi.com</a> SIP/2.0</span></p>



<p><span style="font-size: 10pt; color: navy;">Above are 2 &quot;Invite&quot; request-line examples. In example
one, it indicates a port number, 5555. What does the port number mean?</span><span style="font-size: 10pt; color: navy;"> Is it the port number of SIP proxy?</span></p></div></div></blockquote><div>As you suspected yourself and as Shmulik pointed correctly, the number 555 indicates <br>

the port number of the running SIP Proxy which the UA is registered to.<br></div><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="red" lang="EN-US">

<div><p><span style="font-size: 10pt; color: navy;"> </span></p>



<p><span style="font-size: 10pt; color: navy;">Why we include port number
here? The SIP proxy(<a href="http://biloxi.com" target="_blank">biloxi.com</a>) application layer who receives</span></p></div></div></blockquote><div>Simply because this is a non-standard SIP port, just as if a web server was running on port<br>

8081 you would have to explicitly tell the browser to use that port, for example:<br><a href="http://www.non-standard-server.com:8081" target="_blank">http://www.non-standard-server.com:8081</a><br><br>This is also required when more than one SIP Proxy is involved in the process and the<br>

INVITE travels from one proxy to another. <br><br></div><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="red" lang="EN-US">

<div>

<p><span style="font-size: 10pt; color: navy;">&nbsp;The second &quot;Invite&quot; request-line example is from
RFC3261 and does not include port number. Can I say the port number is useless?
So why example one includes a useless part in &quot;Invite&quot; request-line
URL?</span><br></p></div></div></blockquote><div>To be more precise the first invite message <span style="font-size: 10pt; color: navy;">(INVITE <a href="mailto:sip%3Abob@biloxi.com" target="_blank">sip:bob@biloxi.com</a> SIP/2.0:5555)<br>


</span>is wrong. It should have been INVITE <a href="http://sip:bob@biloxi.com:5555" target="_blank">sip:bob@biloxi.com:5555</a> SIP/2.0&nbsp;</div><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="red" lang="EN-US"><div><p><span style="font-size: 10pt; color: navy;"> </span></p><span style="font-size: 10pt; color: navy;">Any standard on this port number issue? Thanks.</span></div></div></blockquote>

<div><br>It is very common for different SIP stacks to behave somewhat differently and quite<br>a few of them do not comply with the RFC which is very unfortunate.<br></div></div><br><br><br clear="all"><br>-- <br>-- <br>

Regards,<br>
<div dir="ltr" align="left">

<p><span style="font-size: 11pt; font-family: Arial;"><b>Liran 
Tal<br></b></span><span style="font-size: 9pt; font-family: Arial;"><b>Unix Kernel 
Developer<br><font color="#bd3a30"><span>NGN Solutions, 
</span><span>TELES</span>&nbsp;</font><b><span style="font-size: 9pt; font-family: Arial;"><span style="color: rgb(189, 58, 48);">Israel<br><br></span></span></b></b></span><span style="font-size: 9pt; font-family: Arial;"><span></span></span></p>


<p><span style="font-size: 9pt; font-family: Arial;"><span><font size="2"><img alt="[Teles Logo]" src="?ui=2&amp;ik=c9e71119a9&amp;realattid=0.1&amp;attid=0.1&amp;disp=emb&amp;view=att&amp;th=11a5b01e60f118d7" align="bottom" border="0" hspace="0"></font></span></span></p>

<p><span style="font-size: 9pt; font-family: Arial;"><span><font size="2">Phone&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; +972 4 
9092592<br>Fax&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
+972 4 
9597144<br>Mobile&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
+972 52 3793446</font></span></span></p></div>

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Subject: Re: [Sip] [SIPForum-discussion] What is the port number in
	"Invite"	request-line? Thanks
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Stevan hi,

 

The reason is that the proxy is listening to port 5555.

By example one it reference to send the SIP message to this port (port
5555).

By using example two, the SIP message will be sent to the default SIP port
(i.e. 5060)

 

 

 

Sincerely,

* 

Sam Basan

 

From: discussion-bounces@sipforum.org
[mailto:discussion-bounces@sipforum.org] On Behalf Of Yuantao Zhang
Sent: Wednesday, June 04, 2008 12:10
To: sip@ietf.org; discussion@sipforum.org
Subject: [SIPForum-discussion] What is the port number in "Invite"
request-line? Thanks

 

Dear all

 

INVITE sip:bob@biloxi.com SIP/2.0:5555

INVITE sip:bob@biloxi.com SIP/2.0

 

Above are 2 "Invite" request-line examples. In example one, it indicates a
port number, 5555. What does the port number mean? 

 

Is it the port number of SIP proxy? Why we include port number here? The SIP
proxy(biloxi.com) application layer who receives this "Invite" should
receive this via port 5555 from lowe layer(UDP). Therefore, The SIP
proxy(biloxi.com) application layer already know the port is 5555. Actualy,
the SIP proxy application layer keeps listening port 5555. 

 

The second "Invite" request-line example is from RFC3261 and does not
include port number. Can I say the port number is useless? So why example
one includes a useless part in "Invite" request-line URL? 

 

Or is the port number is which the UAS is listening to? How the UAC knows it
is 5555 if so? 

 

Any standard on this port number issue? Thanks. 

 

Best regards

Steven



__________ Information from ESET Smart Security, version of virus signature
database 3157 (20080604) __________

The message was checked by ESET Smart Security.

http://www.eset.com


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

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

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>The reason is that the proxy is listening to port =
5555.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>By example one it reference to send the SIP message to =
this port
(port 5555).<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>By using example two, the SIP message will be sent to the
default SIP port (i.e. 5060)<o:p></o:p></span></p>

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

<div>

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

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

<p class=3DMsoNormal><b><span =
style=3D'font-size:14.0pt;font-family:"Brush Script MT";
color:#003366'>Sincerely,<o:p></o:p></span></b></p>

<p class=3DMsoNormal><span =
style=3D'font-size:14.0pt;font-family:"Arial","sans-serif";
color:black'><img width=3D35 height=3D25 id=3D"Picture_x0020_1"
src=3D"cid:image001.jpg@01C8C741.EF254D20" alt=3D"*">&nbsp;</span><span
style=3D'color:black'><o:p></o:p></span></p>

<p class=3DMsoNormal><b><span =
style=3D'font-size:14.0pt;font-family:"Brush Script MT";
color:#003366'>Sam Basan</span></b><span =
style=3D'font-size:11.0pt;font-family:
"Calibri","sans-serif";color:#1F497D'><o:p></o:p></span></p>

</div>

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

<div>

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

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
discussion-bounces@sipforum.org [mailto:discussion-bounces@sipforum.org] =
<b>On
Behalf Of </b>Yuantao Zhang<br>
<b>Sent:</b> Wednesday, June 04, 2008 12:10<br>
<b>To:</b> sip@ietf.org; discussion@sipforum.org<br>
<b>Subject:</b> [SIPForum-discussion] What is the port number in
&quot;Invite&quot; request-line? Thanks<o:p></o:p></span></p>

</div>

</div>

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

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

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

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>INVITE sip:bob@biloxi.com SIP/2.0:5555<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>INVITE sip:bob@biloxi.com SIP/2.0<o:p></o:p></span></p>

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

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>Above are 2 &#8220;Invite&#8221; request-line examples. In =
example
one, it indicates a port number, 5555. What does the port number mean? =
<o:p></o:p></span></p>

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

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>Is it the port number of SIP proxy? Why we include port =
number
here? The SIP proxy(biloxi.com) application layer who receives this
&quot;Invite&quot; should receive this via port 5555 from lowe =
layer(UDP).
Therefore, The SIP proxy(biloxi.com) application layer already know the =
port is
5555. Actualy, the SIP proxy application layer keeps listening port =
5555. <o:p></o:p></span></p>

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

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>The second &#8220;Invite&#8221; request-line example is from
RFC3261 and does not include port number. Can I say the port number is =
useless?
So why example one includes a useless part in &#8220;Invite&#8221; =
request-line
URL? <o:p></o:p></span></p>

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

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>Or is the port number is which the UAS is listening to? How =
the UAC
knows it is 5555 if so? <o:p></o:p></span></p>

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

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>Any standard on this port number issue? Thanks. =
<o:p></o:p></span></p>

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

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

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

<p class=3DMsoNormal><br>
<br>
__________ Information from ESET Smart Security, version of virus =
signature
database 3157 (20080604) __________<br>
<br>
The message was checked by ESET Smart Security.<br>
<br>
<a href=3D"http://www.eset.com">http://www.eset.com</a><o:p></o:p></p>

</div>

 <BR><BR>__________ Information from ESET Smart Security, version of =
virus signature database 3161 (20080605) __________<BR><BR>The message =
was checked by ESET Smart Security.<BR><BR><A =
HREF=3D"http://www.eset.com">http://www.eset.com</A><BR> </body>

</html>

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_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============1032173765==--



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ok, I stand corrected.

	Thanks,
	Paul

Dean Willis wrote:
> 
> On Jun 11, 2008, at 5:27 PM, Paul Kyzivat wrote:
> 
>>
>>
>> Dale.Worley@comcast.net wrote:
>>>   From: <barcaroller@sympatico.ca>
>>>
>>>   I have a basic SIP question.  The way I understood RFC3261, the 
>>> "To:" header
>>>   field in a response  has to match the "To:" header field of the
>>>   corresponding request (if no To tag is present in the request).  
>>> This is
>>>   true for 99% of the SIP conversations I've seen but not for the 
>>> (real-life)
>>>   example below (note that the "To:" header fields differ).  
>>> Wireshark has no
>>>   problem with analyzing this SIP conversation, so I suspect I'm
>>>   misinterpreting the RFC.
>>>
>>>   Can the "To:" header fields be different, and if so, how can the
>>>   response be correlated to the request?
>>>
>>> The RFC does require the To (and From) headers to match, but everybody
>>> believes that is an obsolete requirement and will probably be relaxed
>>> in the near future.  So generate requests with matching To/From
>>> headers, but do not depend on them.
>>
>> AFAIK there is no agreement that they shouldn't match in a request and
>> its *response*. Most do now agree that they might change for subsequent
>> requests within a dialog.
>>
>> At one point there may have been a little talk about changing the To in
>> a response to indicate the actual identity of the called party. But once
>> it was realized there was no good way to authenticate that we stopped
>> talking about it. (At least that is what I remember.)
>>
> 
> Changing it in a response is critical to extending RFC 4474 for use in 
> responses where there are certificates available in the UAs or where 
> identity-based signatures are used.
> 
> In other words, IF we solve the PKI problem and don't get distracted by 
> the unanticipated respondent problem, then we NEED variant To in 
> responses to be able to se response-identity, because connected-identity 
> is a) late in a dialog and b) useless for non-dialog transactions, like 
> MESSAGE.
> 
> -- 
> Dean
> 
> 
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This situation is known as "glare". It is discussed in rfc 3261 and in 
http://www.ietf.org/internet-drafts/draft-ietf-sipping-race-examples-05.txt.
In short, both sides should send a 491 response, back off, and retry.

	Paul

Maciej Kowalski wrote:
> Hi Everybody.
> 
> I'm currently working on SIP/TDM Gateway application.
> I've encountered following problem.
> During Sip dialog both sides at the exactly same moment send Re-Invite 
> requests.
> Those are reaching their destinations while both client are expecting 
> replies. What should be the behavior of SIP client in a situation like that.
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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Subject: Re: [Sip] PTT query
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atul garg wrote:
> Hello,
> 
> I have a question regarding push to talk message .
> 
> Is there any way through which I can identify that particular SIP message is meant for PTT ie. this sip message is a part of some Push To Talk session..
> 
> Like through which I can filter that such message is for PTT.
> 
> Such criteria which can be applicable on all SIP messages ie.
> 
> 1) Invite
> 2) Register
> 3) 2xx/1xx/4xx/3xx etc. responces
> ..
> ..
> etc.

Are you talking specifically about the 3gpp IMS form of PTT? Or is your 
question more general?

If you are talking about IMS PTT then you should be asking the IMS 
community, because their soluitons are very unique to their environment.

Beyond that, AFAIK there is no specification for how PTT should work.

IMO the concept of PTT is about the characteristics of a media session. 
I don't see why it has anything to do with REGISTER, etc. aside possibly 
the communication of capabilities, which can be handled via callerprefs 
/ callee capabilities.

	Thanks,
	Paul
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From: "Schmidt, Christian 1. (NSN - DE/Muenich)" <christian.1.schmidt@nsn.com>
To: "ext Stuart Hoggan" <s.hoggan@CableLabs.com>,
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Hi,

There is a small error in the link, it should be:
http://www.ietf.org/internet-drafts/draft-dotson-sip-mutual-auth-03.txt

It is useful and it should become SIP WG item.

Regards
Christian




-----Urspr=FCngliche Nachricht-----
Von: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] Im Auftrag von ext =
Stuart Hoggan
Gesendet: Mittwoch, 11. Juni 2008 00:01
An: DRAGE, Keith (Keith); sip@ietf.org
Betreff: Re: [Sip] draft-dotson-sip-mutual-auth-02

All,

We have updated the document to incorporate feedback received so far.
It now contains a section that helps clarify the motivation.  Let us
know if you have further comments. =


-Stuart

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> DRAGE, Keith (Keith)
> Sent: Wednesday, June 04, 2008 3:48 AM
> To: sip@ietf.org
> Subject: Re: [Sip] draft-dotson-sip-mutual-auth-02
> =

> (As SIP WG cochair)
> =

> This has been raised in the SIP group a couple of times, and we have
not
> yet gained an idea of whether to proceed with it in the SIP WG or not.
> =

> There have been a couple of technical comments raised in the past from
> the security experts; my understanding is that these have now been
> clarified.
> =

> I believe in the past there has been an interested community that says
> this is useful. It has also been clarified that 3GPP would like to use
> it in their specifications.
> =

> So could I ask the WG to look at this document, and formally to
indicate
> whether they believe some work with this scope should be progressed as
a
> SIP WG item.
> =

>
http://www.ietf.org/internet-drafts/draft-dotson-sip-mutual-auth-02.txt
> =

> So if you support it please indicate to the SIP WG chairs.
> =

> If you have technical concerns (i.e. it should not proceed because it
is
> technically flawed beyond fixing) then please ideally mail the list,
> although the SIP WG chairs will also accept input.
> =

> Responses please in 1 calendar week from this date.
> =

> Regards
> =

> Keith
> =

> =

> ________________________________
> =

> 	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
> Behalf Of Stuart Hoggan
> 	Sent: Friday, May 02, 2008 5:15 PM
> 	To: sip@ietf.org
> 	Subject: [Sip] draft-dotson-sip-mutual-auth-02
> =

> =

> =

> 	Folks,
> =

> =

> =

> 	Based on the feedback received in Philadelphia, we have updated
> draft-dotson-sip-mutual-auth. The changes clarify the usage of the
> Proxy-Authentication-Info header, including support for multiple
values.
> =

> =

> =

> =

> 	Any further comments or suggestions will be appreciated. It
> would also be nice to obtain WG feedback on pursuing this I-D as a WG
> document.
> =

> =

> =

> 	Regards,
> =

> 	Stuart
> =

> =

> =

> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
_______________________________________________
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This list is for NEW development of the core SIP 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: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
To: <sip@ietf.org>
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Subject: Re: [Sip] draft-dotson-sip-mutual-auth-03
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(As SIP WG cochair)

The document has now been updated based on comments received.

http://www.ietf.org/internet-drafts/draft-dotson-sip-mutual-auth-03.txt

I will give a few more days for responses to the question of support /
no support.

Please read the draft and indicate your postion.

Regards

Keith 

> -----Original Message-----
> From: Stuart Hoggan [mailto:s.hoggan@CableLabs.com] 
> Sent: Tuesday, June 10, 2008 11:01 PM
> To: DRAGE, Keith (Keith); sip@ietf.org
> Subject: RE: [Sip] draft-dotson-sip-mutual-auth-02
> 
> All,
> 
> We have updated the document to incorporate feedback received so far.
> It now contains a section that helps clarify the motivation.  
> Let us know if you have further comments. 
> 
> -Stuart
> 
> > -----Original Message-----
> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of 
> > DRAGE, Keith (Keith)
> > Sent: Wednesday, June 04, 2008 3:48 AM
> > To: sip@ietf.org
> > Subject: Re: [Sip] draft-dotson-sip-mutual-auth-02
> > 
> > (As SIP WG cochair)
> > 
> > This has been raised in the SIP group a couple of times, and we have
> not
> > yet gained an idea of whether to proceed with it in the SIP 
> WG or not.
> > 
> > There have been a couple of technical comments raised in 
> the past from 
> > the security experts; my understanding is that these have now been 
> > clarified.
> > 
> > I believe in the past there has been an interested 
> community that says 
> > this is useful. It has also been clarified that 3GPP would 
> like to use 
> > it in their specifications.
> > 
> > So could I ask the WG to look at this document, and formally to
> indicate
> > whether they believe some work with this scope should be 
> progressed as
> a
> > SIP WG item.
> > 
> >
> http://www.ietf.org/internet-drafts/draft-dotson-sip-mutual-au
> th-02.txt
> > 
> > So if you support it please indicate to the SIP WG chairs.
> > 
> > If you have technical concerns (i.e. it should not proceed 
> because it
> is
> > technically flawed beyond fixing) then please ideally mail 
> the list, 
> > although the SIP WG chairs will also accept input.
> > 
> > Responses please in 1 calendar week from this date.
> > 
> > Regards
> > 
> > Keith
> > 
> > 
> > ________________________________
> > 
> > 	From: sip-bounces@ietf.org 
> [mailto:sip-bounces@ietf.org] On Behalf Of 
> > Stuart Hoggan
> > 	Sent: Friday, May 02, 2008 5:15 PM
> > 	To: sip@ietf.org
> > 	Subject: [Sip] draft-dotson-sip-mutual-auth-02
> > 
> > 
> > 
> > 	Folks,
> > 
> > 
> > 
> > 	Based on the feedback received in Philadelphia, we have updated 
> > draft-dotson-sip-mutual-auth. The changes clarify the usage of the 
> > Proxy-Authentication-Info header, including support for multiple
> values.
> > 
> > 
> > 
> > 
> > 	Any further comments or suggestions will be 
> appreciated. It would 
> > also be nice to obtain WG feedback on pursuing this I-D as a WG 
> > document.
> > 
> > 
> > 
> > 	Regards,
> > 
> > 	Stuart
> > 
> > 
> > 
> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use 
> > sip-implementors@cs.columbia.edu for questions on current sip Use 
> > sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
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From sip-bounces@ietf.org  Thu Jun 12 07:29:09 2008
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Subject: Re: [Sip] Both sides of dialog sending request at the same time
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   From: Maciej Kowalski <macius_13@o2.pl>

   I'm currently working on SIP/TDM Gateway application.
   I've encountered following problem.
   During Sip dialog both sides at the exactly same moment send Re-Invite 
   requests.
   Those are reaching their destinations while both client are expecting 
   replies. What should be the behavior of SIP client in a situation like that.

Sections 14.1 and 14.2 of RFC 3261 give detailed instructions on how
this situation must be handled.

Dale
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Subject: Re: [Sip] PTT query
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If you are talking about OMA PoC (PTT over Cellular), there is a way
defined in OMA standards to do this. It specifies a feature-tag in
contact header to identify PoC calls.

However, this is outside the purview of SIP.

Regards
Satya T
-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Paul Kyzivat
Sent: Thursday, June 12, 2008 6:24 PM
To: shivalk_80@yahoo.co.in
Cc: sip@ietf.org
Subject: Re: [Sip] PTT query



atul garg wrote:
> Hello,
> 
> I have a question regarding push to talk message .
> 
> Is there any way through which I can identify that particular SIP
message is meant for PTT ie. this sip message is a part of some Push To
Talk session..
> 
> Like through which I can filter that such message is for PTT.
> 
> Such criteria which can be applicable on all SIP messages ie.
> 
> 1) Invite
> 2) Register
> 3) 2xx/1xx/4xx/3xx etc. responces
> ..
> ..
> etc.

Are you talking specifically about the 3gpp IMS form of PTT? Or is your
question more general?

If you are talking about IMS PTT then you should be asking the IMS
community, because their soluitons are very unique to their environment.

Beyond that, AFAIK there is no specification for how PTT should work.

IMO the concept of PTT is about the characteristics of a media session. 
I don't see why it has anything to do with REGISTER, etc. aside possibly
the communication of capabilities, which can be handled via callerprefs
/ callee capabilities.

	Thanks,
	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
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_______________________________________________
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On Jun 12, 2008, at 5:58 AM, atul garg wrote:

> Hello,
>
> I have a question regarding push to talk message .
>
> Is there any way through which I can identify that particular SIP  
> message is meant for PTT ie. this sip message is a part of some Push  
> To Talk session..
>
> Like through which I can filter that such message is for PTT.
>
> Such criteria which can be applicable on all SIP messages ie.
>
> 1) Invite
> 2) Register
> 3) 2xx/1xx/4xx/3xx etc. responces

I believe that requires reading the PTT specification. There is no  
IETF PTT specification.

You might want to look at OMA's publicly-available specs:

http://member.openmobilealliance.org/ftp/Public_documents/POC

--
Dean


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Subject: [Sip] Questions regarding SDPs and dialogs...
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Thanks to all who answered my request/response question.  I have a couple 
more of SIP questions.

-1-
I've seen SIP calls where the callee's SDP is transmitted not through the 
usual 200 OK response but through other responses (like 183 Session 
Progress).  In fact, the callee starts sending/receiving RTP/RTCP streams 
long before he/she sends out the 200 OK response.  Is this legit?

-2-
Are RTP/RTCP streams associated with dialogs or with sessions?  In other 
words, assuming a SIP session with multiple dialogs, can each dialog have 
its own (private) set of RTP/RTCP streams?

_______________________________________________
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Subject: Re: [Sip] Questions regarding SDPs and dialogs...
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> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of barcaroller@sympatico.ca
> Sent: Thursday, June 12, 2008 4:01 PM
> To: sip@ietf.org
> Subject: [Sip] Questions regarding SDPs and dialogs...
> 
> Thanks to all who answered my request/response question.  I 
> have a couple 
> more of SIP questions.
> 
> -1-
> I've seen SIP calls where the callee's SDP is transmitted not 
> through the 
> usual 200 OK response but through other responses (like 183 Session 
> Progress).  In fact, the callee starts sending/receiving 
> RTP/RTCP streams 
> long before he/she sends out the 200 OK response.  Is this legit?

Yes, it's called "early media".  One of the better discussions of all sorts of
cases where this occurs is in
http://tools.ietf.org/html/draft-stucker-sipping-early-media-coping-03
(expired).

-d


> -2-
> Are RTP/RTCP streams associated with dialogs or with 
> sessions?  In other 
> words, assuming a SIP session with multiple dialogs, can each 
> dialog have 
> its own (private) set of RTP/RTCP streams?
> 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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


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Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
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Here are some comments on this I-D.  I don't think that any of them
modify our understanding of the proposal.  I guess this is very terse;
please use these suggestions as you see fit.

* Abstract

Clarify:

   This specification defines a new SIP response code, 199 Early
   Dialog Terminated, which a SIP entity can use to indicate upstream
   that an early dialog has been terminated.  The response code can be
   used by a SIP entity (a SIP proxy or UAS), to inform the UAC that
   an early dialog has been terminated.  As the 199 response is a
   provisional response, often the UAC will receive it before
   receiving a final response to its request.

* Section 1

   When a forking proxy receives a non-2xx final response, it does not
   always immediately forward the response upstream towards the sender
   of the associated request.  Instead, the forking proxy "stores" it
   and waits for further final responses from remote destinations where
   the forked request was forwarded.

The text does not allow for serial forking, only parallel forking,
which might be confusing to people who are more aware of serial
forking.  So amend the last sentence to:

   Instead, the forking proxy "stores" it and waits for further final
   responses from remote destinations where the forked request was
   forwarded, and possibly forwards the request to additional
   destinations.

Also, Avoid using "..." except when indicating that the word is used
in a non-standard way or that one is discussing the word itself:

   Instead, the forking proxy "stores" it [...]

should be:

   Instead, the forking proxy stores it [...]

* Section 3.

   REQ 1: It must be possible to indicate to the UAC that an early
   dialog has been terminated before a final response is sent.

Oddly, though everyone thinks they understand this sentence, it
doesn't really say what we want.  To really ensure that the UAC
understands the early dialog has been terminated before the UAS sends
a final response, the UAS would have to send 199 *and get the PRACK
saying that it was received* before sending a final response.  This may
well be worse than the present state of affairs.

What we really mean is:

   REQ 1: It must be possible for a UAS or proxy to send an indication
   to the UAC that an early dialog has been terminated, which
   indication will not be delayed waiting for a final response from
   any UAS.

* Section 4

   When a client receives a 199 Early Dialog Terminated provisional
   response it MAY release resources and procedures associated with the
   early dialog the 199 response is received on.

Amend to clarify the meaning of 199 independently of the consequences
of that meaning:

   When a client receives a 199 Early Dialog Terminated provisional
   response, it has positive knowledge that the UAS that created the
   early dialog has terminated it, and so the client MAY release
   resources and procedures associated with the early dialog for which
   the 199 response is received.

* Section 4

Add this paragraph before the last paragraph:

   A UAC that receives a 199 response for an early dialog MUST NOT
   send any further requests on that dialog, except for a PRACK if the
   199 response was sent reliably, and PRACKs for other reliable
   provisional responses that are to be acknowledged.

This is copied from section 7 (Backward Compatibility), but since it
is a constraint on UACs, it should also be stated in this section.

* Section 4

   The UAC MAY use additional information (for example the final
   response which triggered the 199 response) received in the 199
   response when initiating new sessions, if it is possible to avoid the
   new session initiation request to be rejected.

What is the meaning of the final clause?

* Section 5

Add "early" as marked:

   When a UAS wants to terminate *an early* dialog it sends a non-2xx
   SIP final response, as specified in [RFC3261].  In addition, prior
   to sending the non-2xx SIP final response, the UAS MAY send a 199
   response to indicate that the dialog is being terminated.

This clarifies that 199 cannot be used to terminate a confirmed
dialog.

(The reason a UAS may want to send a 199 is to provide this
information even when the upstream proxy does not support 199.)

There is a subtlty to the second sentence:

Sending the 199 *before* the final response is not necessary due to
the semantics of 199 (as it will probably reach the UAC first anyway,
and it would not matter if it did not), but rather to prevent the
upstream proxy from generating a 199 based on the final response, and
then to transmit the UAS's 199 immediately after -- if the UAS sends
the 199 first, any upstream proxy will (most likely) see the 199
before seeing the final response (which might induce the proxy to send
its own 199).

* Section 6

   When a forking proxy receives a non-2xx final response which
   terminates one or more early dialogs and the proxy does not intend to
   forward the final response immediately (due to the rules for a
   forking proxy) it MUST send a 199 provisional response, for each
   associated early dialog that it can associate with the final
   response, upstream towards the sender of the associated request.

Clarify, and note that if the proxy has already passed a 199 for an
early dialog, it need not generate one itself.  Also, I've reduced
MUST to SHOULD, because (as the following NOTE explains), the proxy
may not know of all early dialogs which are terminated by the final
response.

   [...] it SHOULD initiate a 199 provisional response upstream
   towards the sender of the associated request, for each terminated
   early dialog, excepting those for which it has already transmitted
   a 199 response.

Note that the "already transmitted 199 responses" may have come from
downstream proxies as well as downstream UASs, so this consideration
applies even if we do not allow UASs to send 199s.

* Section 7

   Since all SIP entities involved in a session setup do not necessarily
   support the specific meaning of the 199 Early Dialog Terminated
   provisional response, the sender of the response MUST be prepared to
   receive SIP requests associated with the dialog for which the 199
   response was sent.

This situation can also result due to race conditions.  Amend to:

   Because of race conditions and because the client may not support the
   specific meaning of the 199 provisional response, the sender of the
   response MUST be prepared to receive SIP requests associated with
   the dialog for which the 199 response was sent.

And add a qualifier to this sentence:

   If such request is received, and the receiver maintains state of
   the dialogs, the receiver MUST reply to such requests with a 481
   final response (unless another error response is more appropriate).

* Section 7

Remove "..." from this sentence and add "even" as marked:

   The 199 Early Dialog Terminated response code MUST NOT replace a
   final response.  A final response MUST always be sent, *even* after
   one or many 199 Early Dialog Terminated provisional responses have
   been sent.

* Section 8

Clarify this sentence to:

   The 199 Early Dialog Terminated response code allows a SIP entity
   to indicate to upstream entities that a specific early dialog has
   been terminated, before a final response reaches the upstream
   entities.

Add the following:

   The early dialog that has been terminated is identified by the Call-Id
   header and the to-tag value of the 199 response.

   Future standards may define optional additional information carried
   in the headers and/or body of the response.

* Section 9

   A 199 Early Dialog Terminated provisional response MUST NOT contain a
   new SDP offer/answer message body, but the sender of the response MAY
   insert a copy of a previously sent offer/answer message body as
   otherwise allowed by the offer/answer rules for a provisional
   response.

This rule seems excessively strict, as it seems OK to send a new SDP
answer in a 199, even if the dialog will be immediately terminated.
Certainly, the UAC cannot tell the difference, as there could have
been a previous but lost 1xx response carrying the same SDP.

The interaction of 199 and SDP probably needs further study...

* Section 11

Nit:  Use "reject" here rather than "rejects":

   UAS_2 and UAS_3 *reject* the INVITE by sending a 4xx error response.

Here is a modification of the example, showing how a UAS can send a
199, and how the proxy handles it.  (You may or may not want to use
this.)  I've also added the ACK's for the 4xx's, which were missing in
the original.

   [...] UAS_2 and UAS_3 reject the INVITE by sending a 4xx error
   response.  UAS_3 sends a 199 with its 4xx.  When P1 receives the
   4xx response from UAS_2 it immediately sends a 199 Early Dialog
   Terminated associated with UAS_2's dialog towards the UAC.  P1
   transmits UAS_3's 199, and so does not initiate one itself for
   UAS_3's dialog when it receives UAS_3's 4xx response.

      UAC               P1                  UAS_2    UAS_3    UAS_4

       --- INVITE ------>
                         --- INVITE (leg 2) ->
                         --- INVITE (leg 3) ---------->
                         --- INVITE (leg 4) ------------------->
                         <-- 18x (leg 2) -----
       <-- 18x (leg 2) --
                         <-- 18x (leg 3) --------------
       <-- 18x (leg 3) --
                         <-- 18x (leg 4) -----------------------
       <-- 18x (leg 4) --
                         <-- 4xx (leg 2) -----
       <-- 199 (leg 2) --
                         --- ACK (leg 2) ---->
                         <-- 199 (leg 3) --------------
       <-- 199 (leg 3) --
                         <-- 4xx (leg 3) --------------
                         --- ACK (leg 3) ------------->
                         <-- 200 (leg 4) -----------------------
       <-- 200 (leg 4) --
       --- ACK (leg 4) ->
                         --- ACK (leg 4) ---------------------->

Dale
_______________________________________________
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From: Dale.Worley@comcast.net
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Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt
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   From: "Francois Audet" <audet@nortel.com>

   I really think that this draft should make the inclusion of the original 
   non-2XX final response in the 199 MANDATORY by the forking proxy. This would
   be in a message/sip body as described in sectin 23.4/RFC 3261.

(This is outside the scope of the I-D itself.)

If we envision having the 199 carry information from the associated
failure response, I suggest we use a message/sipfrag body excerpted
from the failure response, rather than copying headers from the
failure response into the 199's headers, or copying the entire failure
response as the body of the 199.

The syntax of sipfrag is unambiguous but extremely flexible (RFC
3420):

           sipfrag = [ start-line ]
                     *message-header
                     [ CRLF [ message-body ] ]

Essentially, it is any subset of the headers of the response,
optionally followed by the body.  This subsetting capability allows
small responses (e.g., omit the Via's, Call-Id, To, From,
Record-Route's, etc.), as well as making topology-hiding
straightforward.

It's also inherently extensible, in that if a proxy/UAS decides to add
more headers to the sipfrag, the UAC knows how to identify and/or
ignore them.

It's also terse, in that carrying any header from the response in a
message/sipfrag body is no longer than carrying it as a header of the
199 response.

Dale
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From sip-bounces@ietf.org  Thu Jun 12 20:23:36 2008
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Date: Thu, 12 Jun 2008 20:23:29 -0700
From: "Francois Audet" <audet@nortel.com>
To: <Dale.Worley@comcast.net>, SIP IETF <sip@ietf.org>
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And do you think we should specify what are the minimum headers thatshould
be carried?

What about MIME bodies?

There might be cases where they are really important, no?

I'm a little worried that the proxy may remove important information.
Especially when we extend SIP.

Maybe instead we should list the removable headers (e.g., all the ones that
are identical to their counterpart in the 199 itself), and then explain
"don't remove what you don't understand"?

Opinions?


On Jun12 2008 20:05 , "Dale.Worley@comcast.net" <Dale.Worley@comcast.net>
wrote:

>    From: "Francois Audet" <audet@nortel.com>
> 
>    I really think that this draft should make the inclusion of the original
>    non-2XX final response in the 199 MANDATORY by the forking proxy. This
> would
>    be in a message/sip body as described in sectin 23.4/RFC 3261.
> 
> (This is outside the scope of the I-D itself.)
> 
> If we envision having the 199 carry information from the associated
> failure response, I suggest we use a message/sipfrag body excerpted
> from the failure response, rather than copying headers from the
> failure response into the 199's headers, or copying the entire failure
> response as the body of the 199.
> 
> The syntax of sipfrag is unambiguous but extremely flexible (RFC
> 3420):
> 
>            sipfrag = [ start-line ]
>                      *message-header
>                      [ CRLF [ message-body ] ]
> 
> Essentially, it is any subset of the headers of the response,
> optionally followed by the body.  This subsetting capability allows
> small responses (e.g., omit the Via's, Call-Id, To, From,
> Record-Route's, etc.), as well as making topology-hiding
> straightforward.
> 
> It's also inherently extensible, in that if a proxy/UAS decides to add
> more headers to the sipfrag, the UAC knows how to identify and/or
> ignore them.
> 
> It's also terse, in that carrying any header from the response in a
> message/sipfrag body is no longer than carrying it as a header of the
> 199 response.
> 
> Dale
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

_______________________________________________
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From sip-bounces@ietf.org  Thu Jun 12 23:25:06 2008
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Thank You for such quick and detailed response.

Maciej

Maciej Kowalski pisze:
> Hi Everybody.
>
> I'm currently working on SIP/TDM Gateway application.
> I've encountered following problem.
> During Sip dialog both sides at the exactly same moment send Re-Invite 
> requests.
> Those are reaching their destinations while both client are expecting 
> replies. What should be the behavior of SIP client in a situation like that.
>
>
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>
>   

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From sip-bounces@ietf.org  Thu Jun 12 23:41:56 2008
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From: Rockson Zhengyu Li <lzyiii@laszlomail.com>
To: KevinKuei <kkuei@above.net.tw>
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Subject: Re: [Sip] [sip] handling ACK for 2xx response in proxy??
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ACK received from upstream at transport layer would be forwarded to 
proxy core directly,
which would action as per sec 16.


sec 18.2.1

    Next, the server transport attempts to match the request to a server
    transaction.  It does so using the matching rules described in
    Section 17.2.3.  If a matching server transaction is found, the
    request is passed to that transaction for processing.  If no match 
is
    found, the request is passed to the core, which may decide to
    construct a new server transaction for that request.  Note that when
    a UAS core sends a 2xx response to INVITE, the server transaction is
    destroyed.  This means that when the ACK arrives, there will be no
    matching server transaction, and based on this rule, the ACK is
    passed to the UAS core, where it is processed.

Regards,
-Rockson

On Mon, Oct 29, 2007 at 12:06 PM, KevinKuei wrote:

> Hello all,
>
> In RFC3261 sec.16, is there anywhere talking about stateful proxy 
> handling ACK??
> Since ACK for 2xx response is not send thru transaction, it should not 
> be
> received by server transaction in proxy. Is that correct?? then.. how 
> proxy
> core got ACK??
>
>
> --
> Kevin Kuei
>
>
> ----------------------------------------------------------------
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>
> _______________________________________________
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> Use sip-implementors@cs.columbia.edu for questions on current sip
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Hi Atul

Have a look at the OMA PoC spec
http://www.openmobilealliance.org/technical/release_program/docs/PoC/V2_
0-20080507-C/OMA-TS-PoC_ControlPlane-V2_0-20080507-C.pdf to see what
media feature tags are used in SIP Messages for PoC. 


Regards
Ranjit

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
atul garg
Sent: Thursday, June 12, 2008 4:28 PM
To: sip@ietf.org
Subject: [Sip] PTT query

Hello,

I have a question regarding push to talk message .

Is there any way through which I can identify that particular SIP
message is meant for PTT ie. this sip message is a part of some Push To
Talk session..

Like through which I can filter that such message is for PTT.

Such criteria which can be applicable on all SIP messages ie.

1) Invite
2) Register
3) 2xx/1xx/4xx/3xx etc. responces
..
..
etc.

Thnx
Atul



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Hi Dale, 

>Here are some comments on this I-D.  I don't think that any of them
modify our understanding of the proposal.  I guess 
>this is very terse; please use these suggestions as you see fit.
>
>* Abstract
>
>Clarify:
>
>   This specification defines a new SIP response code, 199 Early
>   Dialog Terminated, which a SIP entity can use to indicate upstream
>   that an early dialog has been terminated.  The response code can be
>   used by a SIP entity (a SIP proxy or UAS), to inform the UAC that
>   an early dialog has been terminated.  As the 199 response is a
>   provisional response, often the UAC will receive it before
>   receiving a final response to its request.

[CHH] Ok.


>* Section 1
>
>   When a forking proxy receives a non-2xx final response, it does not
>   always immediately forward the response upstream towards the sender
>   of the associated request.  Instead, the forking proxy "stores" it
>   and waits for further final responses from remote destinations where
>   the forked request was forwarded.
>
>The text does not allow for serial forking, only parallel forking,
which might be confusing to people who are more aware >of serial
forking.  So amend the last sentence to:
>
>   Instead, the forking proxy "stores" it and waits for further final
>   responses from remote destinations where the forked request was
>   forwarded, and possibly forwards the request to additional
>   destinations.

[CHH] I can change the text. My intention was not to prevent serial
forking. Because, no matter what type of forking it is, the forking
proxy will "store" final responses from the different forked legs.


>Also, Avoid using "..." except when indicating that the word is used in
a non-standard way or that one is discussing the >word itself:
>
>   Instead, the forking proxy "stores" it [...]
>
>should be:
>
>   Instead, the forking proxy stores it [...]

[CHH] Ok.


>* Section 3.
>
>   REQ 1: It must be possible to indicate to the UAC that an early
>   dialog has been terminated before a final response is sent.
>
>Oddly, though everyone thinks they understand this sentence, it doesn't
really say what we want.  To really ensure that 
>the UAC understands the early dialog has been terminated before the UAS
sends a final response, the UAS would have to 
>send 199 *and get the PRACK saying that it was received* before sending
a final response.  This may well be worse than 
>the present state of affairs.
>
>What we really mean is:
>
>   REQ 1: It must be possible for a UAS or proxy to send an indication
>   to the UAC that an early dialog has been terminated, which
>   indication will not be delayed waiting for a final response from
>   any UAS.

[CHH] I hear what you are saying, but it is a copy of the requirement
text we agreed to in SIPPING.


>* Section 4
>
>   When a client receives a 199 Early Dialog Terminated provisional
>   response it MAY release resources and procedures associated with the
>   early dialog the 199 response is received on.
>
>Amend to clarify the meaning of 199 independently of the consequences
of that meaning:
>
>   When a client receives a 199 Early Dialog Terminated provisional
>   response, it has positive knowledge that the UAS that created the
>   early dialog has terminated it, and so the client MAY release
>   resources and procedures associated with the early dialog for which
>   the 199 response is received.

[CHH] Ok.


>* Section 4
>
>Add this paragraph before the last paragraph:
>
>   A UAC that receives a 199 response for an early dialog MUST NOT
>   send any further requests on that dialog, except for a PRACK if the
>   199 response was sent reliably, and PRACKs for other reliable
>   provisional responses that are to be acknowledged.
>
>This is copied from section 7 (Backward Compatibility), but since it is
a constraint on UACs, it should also be stated in 
>this section.

[CHH] Whether the text should be in the document at all depends on if we
allow 199 to be sent reliably in the first place. Based on the comments
received so far we should not mandate 199 to be sent reliably, even if
100rel is required by the UAC. But, the question if whether we want to
FORBID sending it reliably.


>* Section 4
>
>   The UAC MAY use additional information (for example the final
>   response which triggered the 199 response) received in the 199
>   response when initiating new sessions, if it is possible to avoid
the
>   new session initiation request to be rejected.
>
>What is the meaning of the final clause?

[CHH] Others have asked the same, so I guess it needs some modification.
The meaning is that the information received in the 199 may be used to
solve the HERFP problem. It is not in the scope of the document to solve
that problem, just to mention that the information may be used for a
future mechanism solving it.


>* Section 5
>
>Add "early" as marked:
>
>  When a UAS wants to terminate *an early* dialog it sends a non-2xx
>   SIP final response, as specified in [RFC3261].  In addition, prior
>  to sending the non-2xx SIP final response, the UAS MAY send a 199
>   response to indicate that the dialog is being terminated.
>
>This clarifies that 199 cannot be used to terminate a confirmed dialog.

[CHH] I have nothing against your proposal as such, but is this marking
method that you are proposing something we normally use in IETF specs?


>(The reason a UAS may want to send a 199 is to provide this information
even when the upstream proxy does not support 
>199.)
>
>There is a subtlty to the second sentence:
>
>Sending the 199 *before* the final response is not necessary due to the
semantics of 199 (as it will probably reach the 
>UAC first anyway, and it would not matter if it did not), but rather to
prevent the upstream proxy from generating a 199 
>based on the final response, and then to transmit the UAS's 199
immediately after -- if the UAS sends the 199 first, any 
>upstream proxy will (most likely) see the 199 before seeing the final
response (which might induce the proxy to send its 
>own 199).

[CHH] This depends on whether we should keep the UAS text in the first
place, or only describe when proxies send 199. Based on the comments
I've received so far, we should not describe UAS behavior for sending
199.


>* Section 6
>
>   When a forking proxy receives a non-2xx final response which
>   terminates one or more early dialogs and the proxy does not intend
to
>   forward the final response immediately (due to the rules for a
>   forking proxy) it MUST send a 199 provisional response, for each
>   associated early dialog that it can associate with the final
>   response, upstream towards the sender of the associated request.
>
>Clarify, and note that if the proxy has already passed a 199 for an
early dialog, it need not generate one itself.  Also, 
>I've reduced MUST to SHOULD, because (as the following NOTE explains),
the proxy may not know of all early dialogs which 
>are terminated by the final response.
>
>   [...] it SHOULD initiate a 199 provisional response upstream
>   towards the sender of the associated request, for each terminated
>   early dialog, excepting those for which it has already transmitted
>   a 199 response.
>
>Note that the "already transmitted 199 responses" may have come from
downstream proxies as well as downstream UASs, so 
>this consideration applies even if we do not allow UASs to send 199s.

[CHH] I need to think a little more, but it looks ok.



>* Section 7
>
>   Since all SIP entities involved in a session setup do not
necessarily
>   support the specific meaning of the 199 Early Dialog Terminated
>   provisional response, the sender of the response MUST be prepared to
>   receive SIP requests associated with the dialog for which the 199
>   response was sent.
>
>This situation can also result due to race conditions.  Amend to:
>
>   Because of race conditions and because the client may not support
the
>   specific meaning of the 199 provisional response, the sender of the
>   response MUST be prepared to receive SIP requests associated with
>   the dialog for which the 199 response was sent.
>
>And add a qualifier to this sentence:
>
>   If such request is received, and the receiver maintains state of
>   the dialogs, the receiver MUST reply to such requests with a 481
>   final response (unless another error response is more appropriate).

[CHH Ok.


>* Section 7
>
>Remove "..." from this sentence and add "even" as marked:
>
>   The 199 Early Dialog Terminated response code MUST NOT replace a
>   final response.  A final response MUST always be sent, *even* after
>   one or many 199 Early Dialog Terminated provisional responses have
>   been sent.

[CHH] Same issue as before regarding the marking method.


>* Section 8
>
>Clarify this sentence to:
>
>   The 199 Early Dialog Terminated response code allows a SIP entity
>   to indicate to upstream entities that a specific early dialog has
>   been terminated, before a final response reaches the upstream
>   entities.
>
>Add the following:
>
>   The early dialog that has been terminated is identified by the
Call-Id
>   header and the to-tag value of the 199 response.
>
>   Future standards may define optional additional information carried
>   in the headers and/or body of the response.

[CHH] I am ok with the first sentence, but I don't understand the
meaning of the other ("Future standards...") in this context.


>* Section 9
>
>   A 199 Early Dialog Terminated provisional response MUST NOT contain
a
>   new SDP offer/answer message body, but the sender of the response
MAY
>   insert a copy of a previously sent offer/answer message body as
>   otherwise allowed by the offer/answer rules for a provisional
>   response.
>
>This rule seems excessively strict, as it seems OK to send a new SDP
answer in a 199, even if the dialog will be 
>immediately terminated.
>Certainly, the UAC cannot tell the difference, as there could have been
a previous but lost 1xx response carrying the 
>same SDP.
>
>The interaction of 199 and SDP probably needs further study...

[CHH] If we are not going to specify UAS sending of 199 then there will
be no SDP in it. I don't think the forking proxy should store received
SDPs, and then insert one when triggering the 199.


>* Section 11
>
>Nit:  Use "reject" here rather than "rejects":
>
>
>   UAS_2 and UAS_3 *reject* the INVITE by sending a 4xx error response.

[CHH] Ok.


>Here is a modification of the example, showing how a UAS can send a
199, and how the proxy handles it.  (You may or may 
>not want to usethis.) I've also added the ACK's for the 4xx's, which
were missing in the original.
>
>   [...] UAS_2 and UAS_3 reject the INVITE by sending a 4xx error
>   response.  UAS_3 sends a 199 with its 4xx.  When P1 receives the
>  4xx response from UAS_2 it immediately sends a 199 Early Dialog
>   Terminated associated with UAS_2's dialog towards the UAC.  P1
>   transmits UAS_3's 199, and so does not initiate one itself for
>   UAS_3's dialog when it receives UAS_3's 4xx response.
>
>      UAC               P1                  UAS_2    UAS_3    UAS_4
>
>       --- INVITE ------>
>                         --- INVITE (leg 2) ->
>                         --- INVITE (leg 3) ---------->
>                         --- INVITE (leg 4) ------------------->
>                         <-- 18x (leg 2) -----
>       <-- 18x (leg 2) --
>                         <-- 18x (leg 3) --------------
>       <-- 18x (leg 3) --
>                         <-- 18x (leg 4) -----------------------
>       <-- 18x (leg 4) --
>                         <-- 4xx (leg 2) -----
>       <-- 199 (leg 2) --
>                         --- ACK (leg 2) ---->
>                         <-- 199 (leg 3) --------------
>       <-- 199 (leg 3) --
>                         <-- 4xx (leg 3) --------------
>                         --- ACK (leg 3) ------------->
>                         <-- 200 (leg 4) -----------------------
>       <-- 200 (leg 4) --
>       --- ACK (leg 4) ->
>                         --- ACK (leg 4) ---------------------->

[CHH] I'm ok with your modficiation of the example.

Thanks!

Regards,

Christer
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Hi,

I assume there aren't going to be that many headers in the non-2xx error re=
sponse, and I think most will be identical to what will be used in the 199 =
anyway - To, From, Call-ID etc.

I am ok with also talking about SIP headers, in addition to response code a=
nd possible SDP body, but I don't think we should make this too complex.

And, to answer Dale, the intention IS to use sipfrag - no matter how much/l=
ittle information of the final response we include in the 199.

Regards,

Christer

 =


-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Franc=
ois Audet
Sent: 13. kes=E4kuuta 2008 6:23
To: Dale.Worley@comcast.net; SIP IETF
Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt

And do you think we should specify what are the minimum headers thatshould =
be carried?

What about MIME bodies?

There might be cases where they are really important, no?

I'm a little worried that the proxy may remove important information.
Especially when we extend SIP.

Maybe instead we should list the removable headers (e.g., all the ones that=
 are identical to their counterpart in the 199 itself), and then explain "d=
on't remove what you don't understand"?

Opinions?


On Jun12 2008 20:05 , "Dale.Worley@comcast.net" <Dale.Worley@comcast.net>
wrote:

>    From: "Francois Audet" <audet@nortel.com>
> =

>    I really think that this draft should make the inclusion of the origin=
al
>    non-2XX final response in the 199 MANDATORY by the forking proxy. =

> This would
>    be in a message/sip body as described in sectin 23.4/RFC 3261.
> =

> (This is outside the scope of the I-D itself.)
> =

> If we envision having the 199 carry information from the associated =

> failure response, I suggest we use a message/sipfrag body excerpted =

> from the failure response, rather than copying headers from the =

> failure response into the 199's headers, or copying the entire failure =

> response as the body of the 199.
> =

> The syntax of sipfrag is unambiguous but extremely flexible (RFC
> 3420):
> =

>            sipfrag =3D [ start-line ]
>                      *message-header
>                      [ CRLF [ message-body ] ]
> =

> Essentially, it is any subset of the headers of the response, =

> optionally followed by the body.  This subsetting capability allows =

> small responses (e.g., omit the Via's, Call-Id, To, From, =

> Record-Route's, etc.), as well as making topology-hiding =

> straightforward.
> =

> It's also inherently extensible, in that if a proxy/UAS decides to add =

> more headers to the sipfrag, the UAC knows how to identify and/or =

> ignore them.
> =

> It's also terse, in that carrying any header from the response in a =

> message/sipfrag body is no longer than carrying it as a header of the
> 199 response.
> =

> Dale
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use =

> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

_______________________________________________
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Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
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This all sounds like good stuff.

However I am still scratching my head about the UAS sending a 199. While 
I see no reason to forbid it I'm trying to find a situation when it 
would be helpful. The only case I have been able to think of is:

   There is a forking proxy upstream of the UAS, but it doesn't
   support 199. Yet it may delay the final response while it
   tries other forks. So the UAS sending the 199 makes up for
   the inadequacy of the forking proxy.

The trouble with this is that it is inefficient in many cases. If each 
proxy that forks also supports the sending of 199, and the UAS doesn't 
send one, then we avoid sending the message except in cases where it is 
helpful. If the UAS always sends it we have increased the message count 
by one for all failing calls. And the majority of failing calls get no 
benefit from a 199.

This is one of those cases that could probably benefit from simulation 
to judge the merits. My gut feel is that the costs of the UAS sending 
the 199 outweigh the benefits. Barring some analysis to the contrary I 
think I would say that a UAS SHOULD NOT send a 199. Or perhaps MAY send, 
but without any guidance of when.

	Paul

Dale.Worley@comcast.net wrote:
> Here are some comments on this I-D.  I don't think that any of them
> modify our understanding of the proposal.  I guess this is very terse;
> please use these suggestions as you see fit.
> 
> * Abstract
> 
> Clarify:
> 
>    This specification defines a new SIP response code, 199 Early
>    Dialog Terminated, which a SIP entity can use to indicate upstream
>    that an early dialog has been terminated.  The response code can be
>    used by a SIP entity (a SIP proxy or UAS), to inform the UAC that
>    an early dialog has been terminated.  As the 199 response is a
>    provisional response, often the UAC will receive it before
>    receiving a final response to its request.
> 
> * Section 1
> 
>    When a forking proxy receives a non-2xx final response, it does not
>    always immediately forward the response upstream towards the sender
>    of the associated request.  Instead, the forking proxy "stores" it
>    and waits for further final responses from remote destinations where
>    the forked request was forwarded.
> 
> The text does not allow for serial forking, only parallel forking,
> which might be confusing to people who are more aware of serial
> forking.  So amend the last sentence to:
> 
>    Instead, the forking proxy "stores" it and waits for further final
>    responses from remote destinations where the forked request was
>    forwarded, and possibly forwards the request to additional
>    destinations.
> 
> Also, Avoid using "..." except when indicating that the word is used
> in a non-standard way or that one is discussing the word itself:
> 
>    Instead, the forking proxy "stores" it [...]
> 
> should be:
> 
>    Instead, the forking proxy stores it [...]
> 
> * Section 3.
> 
>    REQ 1: It must be possible to indicate to the UAC that an early
>    dialog has been terminated before a final response is sent.
> 
> Oddly, though everyone thinks they understand this sentence, it
> doesn't really say what we want.  To really ensure that the UAC
> understands the early dialog has been terminated before the UAS sends
> a final response, the UAS would have to send 199 *and get the PRACK
> saying that it was received* before sending a final response.  This may
> well be worse than the present state of affairs.
> 
> What we really mean is:
> 
>    REQ 1: It must be possible for a UAS or proxy to send an indication
>    to the UAC that an early dialog has been terminated, which
>    indication will not be delayed waiting for a final response from
>    any UAS.
> 
> * Section 4
> 
>    When a client receives a 199 Early Dialog Terminated provisional
>    response it MAY release resources and procedures associated with the
>    early dialog the 199 response is received on.
> 
> Amend to clarify the meaning of 199 independently of the consequences
> of that meaning:
> 
>    When a client receives a 199 Early Dialog Terminated provisional
>    response, it has positive knowledge that the UAS that created the
>    early dialog has terminated it, and so the client MAY release
>    resources and procedures associated with the early dialog for which
>    the 199 response is received.
> 
> * Section 4
> 
> Add this paragraph before the last paragraph:
> 
>    A UAC that receives a 199 response for an early dialog MUST NOT
>    send any further requests on that dialog, except for a PRACK if the
>    199 response was sent reliably, and PRACKs for other reliable
>    provisional responses that are to be acknowledged.
> 
> This is copied from section 7 (Backward Compatibility), but since it
> is a constraint on UACs, it should also be stated in this section.
> 
> * Section 4
> 
>    The UAC MAY use additional information (for example the final
>    response which triggered the 199 response) received in the 199
>    response when initiating new sessions, if it is possible to avoid the
>    new session initiation request to be rejected.
> 
> What is the meaning of the final clause?
> 
> * Section 5
> 
> Add "early" as marked:
> 
>    When a UAS wants to terminate *an early* dialog it sends a non-2xx
>    SIP final response, as specified in [RFC3261].  In addition, prior
>    to sending the non-2xx SIP final response, the UAS MAY send a 199
>    response to indicate that the dialog is being terminated.
> 
> This clarifies that 199 cannot be used to terminate a confirmed
> dialog.
> 
> (The reason a UAS may want to send a 199 is to provide this
> information even when the upstream proxy does not support 199.)
> 
> There is a subtlty to the second sentence:
> 
> Sending the 199 *before* the final response is not necessary due to
> the semantics of 199 (as it will probably reach the UAC first anyway,
> and it would not matter if it did not), but rather to prevent the
> upstream proxy from generating a 199 based on the final response, and
> then to transmit the UAS's 199 immediately after -- if the UAS sends
> the 199 first, any upstream proxy will (most likely) see the 199
> before seeing the final response (which might induce the proxy to send
> its own 199).
> 
> * Section 6
> 
>    When a forking proxy receives a non-2xx final response which
>    terminates one or more early dialogs and the proxy does not intend to
>    forward the final response immediately (due to the rules for a
>    forking proxy) it MUST send a 199 provisional response, for each
>    associated early dialog that it can associate with the final
>    response, upstream towards the sender of the associated request.
> 
> Clarify, and note that if the proxy has already passed a 199 for an
> early dialog, it need not generate one itself.  Also, I've reduced
> MUST to SHOULD, because (as the following NOTE explains), the proxy
> may not know of all early dialogs which are terminated by the final
> response.
> 
>    [...] it SHOULD initiate a 199 provisional response upstream
>    towards the sender of the associated request, for each terminated
>    early dialog, excepting those for which it has already transmitted
>    a 199 response.
> 
> Note that the "already transmitted 199 responses" may have come from
> downstream proxies as well as downstream UASs, so this consideration
> applies even if we do not allow UASs to send 199s.
> 
> * Section 7
> 
>    Since all SIP entities involved in a session setup do not necessarily
>    support the specific meaning of the 199 Early Dialog Terminated
>    provisional response, the sender of the response MUST be prepared to
>    receive SIP requests associated with the dialog for which the 199
>    response was sent.
> 
> This situation can also result due to race conditions.  Amend to:
> 
>    Because of race conditions and because the client may not support the
>    specific meaning of the 199 provisional response, the sender of the
>    response MUST be prepared to receive SIP requests associated with
>    the dialog for which the 199 response was sent.
> 
> And add a qualifier to this sentence:
> 
>    If such request is received, and the receiver maintains state of
>    the dialogs, the receiver MUST reply to such requests with a 481
>    final response (unless another error response is more appropriate).
> 
> * Section 7
> 
> Remove "..." from this sentence and add "even" as marked:
> 
>    The 199 Early Dialog Terminated response code MUST NOT replace a
>    final response.  A final response MUST always be sent, *even* after
>    one or many 199 Early Dialog Terminated provisional responses have
>    been sent.
> 
> * Section 8
> 
> Clarify this sentence to:
> 
>    The 199 Early Dialog Terminated response code allows a SIP entity
>    to indicate to upstream entities that a specific early dialog has
>    been terminated, before a final response reaches the upstream
>    entities.
> 
> Add the following:
> 
>    The early dialog that has been terminated is identified by the Call-Id
>    header and the to-tag value of the 199 response.
> 
>    Future standards may define optional additional information carried
>    in the headers and/or body of the response.
> 
> * Section 9
> 
>    A 199 Early Dialog Terminated provisional response MUST NOT contain a
>    new SDP offer/answer message body, but the sender of the response MAY
>    insert a copy of a previously sent offer/answer message body as
>    otherwise allowed by the offer/answer rules for a provisional
>    response.
> 
> This rule seems excessively strict, as it seems OK to send a new SDP
> answer in a 199, even if the dialog will be immediately terminated.
> Certainly, the UAC cannot tell the difference, as there could have
> been a previous but lost 1xx response carrying the same SDP.
> 
> The interaction of 199 and SDP probably needs further study...
> 
> * Section 11
> 
> Nit:  Use "reject" here rather than "rejects":
> 
>    UAS_2 and UAS_3 *reject* the INVITE by sending a 4xx error response.
> 
> Here is a modification of the example, showing how a UAS can send a
> 199, and how the proxy handles it.  (You may or may not want to use
> this.)  I've also added the ACK's for the 4xx's, which were missing in
> the original.
> 
>    [...] UAS_2 and UAS_3 reject the INVITE by sending a 4xx error
>    response.  UAS_3 sends a 199 with its 4xx.  When P1 receives the
>    4xx response from UAS_2 it immediately sends a 199 Early Dialog
>    Terminated associated with UAS_2's dialog towards the UAC.  P1
>    transmits UAS_3's 199, and so does not initiate one itself for
>    UAS_3's dialog when it receives UAS_3's 4xx response.
> 
>       UAC               P1                  UAS_2    UAS_3    UAS_4
> 
>        --- INVITE ------>
>                          --- INVITE (leg 2) ->
>                          --- INVITE (leg 3) ---------->
>                          --- INVITE (leg 4) ------------------->
>                          <-- 18x (leg 2) -----
>        <-- 18x (leg 2) --
>                          <-- 18x (leg 3) --------------
>        <-- 18x (leg 3) --
>                          <-- 18x (leg 4) -----------------------
>        <-- 18x (leg 4) --
>                          <-- 4xx (leg 2) -----
>        <-- 199 (leg 2) --
>                          --- ACK (leg 2) ---->
>                          <-- 199 (leg 3) --------------
>        <-- 199 (leg 3) --
>                          <-- 4xx (leg 3) --------------
>                          --- ACK (leg 3) ------------->
>                          <-- 200 (leg 4) -----------------------
>        <-- 200 (leg 4) --
>        --- ACK (leg 4) ->
>                          --- ACK (leg 4) ---------------------->
> 
> Dale
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
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Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt
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My first thought was to agree with Dale, and in fact I already suggested 
use of sipfrag. But I already have experience from the refer NOTIFY that 
sipfrag info is difficult to take advantage of if you have no guarantee 
of what will be included.

IIRC this 199 work was explicitly not expected to solve the full HERFP 
problem. Yet if it could serve for that it would be a good thing. Its 
just that thrashing out everything that is needed will take a bunch of time.

Also, sending back the whole message will make more data to send in 
cases where it often isn't useful.

We could allow a body containing *either* message/sip or 
message/sipfrag, and make them both conditional on presence of a 
corresponding Accept header in the request. If there was no Accept 
header for either then no body would be included. If there was an accept 
for both, message/sip would be be preferred. When a sipfrag is returned 
in a 199, we could require that it include at least the header line, and 
the rest would be at the discretion of the proxy.

	Paul

Francois Audet wrote:
> And do you think we should specify what are the minimum headers thatshould
> be carried?
> 
> What about MIME bodies?
> 
> There might be cases where they are really important, no?
> 
> I'm a little worried that the proxy may remove important information.
> Especially when we extend SIP.
> 
> Maybe instead we should list the removable headers (e.g., all the ones that
> are identical to their counterpart in the 199 itself), and then explain
> "don't remove what you don't understand"?
> 
> Opinions?
> 
> 
> On Jun12 2008 20:05 , "Dale.Worley@comcast.net" <Dale.Worley@comcast.net>
> wrote:
> 
>>    From: "Francois Audet" <audet@nortel.com>
>>
>>    I really think that this draft should make the inclusion of the original
>>    non-2XX final response in the 199 MANDATORY by the forking proxy. This
>> would
>>    be in a message/sip body as described in sectin 23.4/RFC 3261.
>>
>> (This is outside the scope of the I-D itself.)
>>
>> If we envision having the 199 carry information from the associated
>> failure response, I suggest we use a message/sipfrag body excerpted
>> from the failure response, rather than copying headers from the
>> failure response into the 199's headers, or copying the entire failure
>> response as the body of the 199.
>>
>> The syntax of sipfrag is unambiguous but extremely flexible (RFC
>> 3420):
>>
>>            sipfrag = [ start-line ]
>>                      *message-header
>>                      [ CRLF [ message-body ] ]
>>
>> Essentially, it is any subset of the headers of the response,
>> optionally followed by the body.  This subsetting capability allows
>> small responses (e.g., omit the Via's, Call-Id, To, From,
>> Record-Route's, etc.), as well as making topology-hiding
>> straightforward.
>>
>> It's also inherently extensible, in that if a proxy/UAS decides to add
>> more headers to the sipfrag, the UAC knows how to identify and/or
>> ignore them.
>>
>> It's also terse, in that carrying any header from the response in a
>> message/sipfrag body is no longer than carrying it as a header of the
>> 199 response.
>>
>> Dale
>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
> 
> _______________________________________________
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Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt
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Paul Kyzivat wrote:

> Also, sending back the whole message will make more data to send in 
> cases where it often isn't useful.

It is also likely to break UDP. See the problems we had with diagnostic 
responses in general.

--
Dean
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From sip-bounces@ietf.org  Fri Jun 13 10:43:11 2008
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From: "Francois Audet" <audet@nortel.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>, <Dale.Worley@comcast.net>
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Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
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I can think of one scenario...

If the UAS in question is actually some sort of B2BUA. 

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of Paul Kyzivat
> Sent: Friday, June 13, 2008 07:07
> To: Dale.Worley@comcast.net
> Cc: sip@ietf.org
> Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
> 
> This all sounds like good stuff.
> 
> However I am still scratching my head about the UAS sending a 
> 199. While I see no reason to forbid it I'm trying to find a 
> situation when it would be helpful. The only case I have been 
> able to think of is:
> 
>    There is a forking proxy upstream of the UAS, but it doesn't
>    support 199. Yet it may delay the final response while it
>    tries other forks. So the UAS sending the 199 makes up for
>    the inadequacy of the forking proxy.
> 
> The trouble with this is that it is inefficient in many 
> cases. If each proxy that forks also supports the sending of 
> 199, and the UAS doesn't send one, then we avoid sending the 
> message except in cases where it is helpful. If the UAS 
> always sends it we have increased the message count by one 
> for all failing calls. And the majority of failing calls get 
> no benefit from a 199.
> 
> This is one of those cases that could probably benefit from 
> simulation to judge the merits. My gut feel is that the costs 
> of the UAS sending the 199 outweigh the benefits. Barring 
> some analysis to the contrary I think I would say that a UAS 
> SHOULD NOT send a 199. Or perhaps MAY send, but without any 
> guidance of when.
> 
> 	Paul
> 
> Dale.Worley@comcast.net wrote:
> > Here are some comments on this I-D.  I don't think that any of them 
> > modify our understanding of the proposal.  I guess this is 
> very terse; 
> > please use these suggestions as you see fit.
> > 
> > * Abstract
> > 
> > Clarify:
> > 
> >    This specification defines a new SIP response code, 199 Early
> >    Dialog Terminated, which a SIP entity can use to 
> indicate upstream
> >    that an early dialog has been terminated.  The response 
> code can be
> >    used by a SIP entity (a SIP proxy or UAS), to inform the UAC that
> >    an early dialog has been terminated.  As the 199 response is a
> >    provisional response, often the UAC will receive it before
> >    receiving a final response to its request.
> > 
> > * Section 1
> > 
> >    When a forking proxy receives a non-2xx final response, 
> it does not
> >    always immediately forward the response upstream towards 
> the sender
> >    of the associated request.  Instead, the forking proxy 
> "stores" it
> >    and waits for further final responses from remote 
> destinations where
> >    the forked request was forwarded.
> > 
> > The text does not allow for serial forking, only parallel forking, 
> > which might be confusing to people who are more aware of serial 
> > forking.  So amend the last sentence to:
> > 
> >    Instead, the forking proxy "stores" it and waits for 
> further final
> >    responses from remote destinations where the forked request was
> >    forwarded, and possibly forwards the request to additional
> >    destinations.
> > 
> > Also, Avoid using "..." except when indicating that the 
> word is used 
> > in a non-standard way or that one is discussing the word itself:
> > 
> >    Instead, the forking proxy "stores" it [...]
> > 
> > should be:
> > 
> >    Instead, the forking proxy stores it [...]
> > 
> > * Section 3.
> > 
> >    REQ 1: It must be possible to indicate to the UAC that an early
> >    dialog has been terminated before a final response is sent.
> > 
> > Oddly, though everyone thinks they understand this sentence, it 
> > doesn't really say what we want.  To really ensure that the UAC 
> > understands the early dialog has been terminated before the 
> UAS sends 
> > a final response, the UAS would have to send 199 *and get the PRACK 
> > saying that it was received* before sending a final response.  This 
> > may well be worse than the present state of affairs.
> > 
> > What we really mean is:
> > 
> >    REQ 1: It must be possible for a UAS or proxy to send an 
> indication
> >    to the UAC that an early dialog has been terminated, which
> >    indication will not be delayed waiting for a final response from
> >    any UAS.
> > 
> > * Section 4
> > 
> >    When a client receives a 199 Early Dialog Terminated provisional
> >    response it MAY release resources and procedures 
> associated with the
> >    early dialog the 199 response is received on.
> > 
> > Amend to clarify the meaning of 199 independently of the 
> consequences 
> > of that meaning:
> > 
> >    When a client receives a 199 Early Dialog Terminated provisional
> >    response, it has positive knowledge that the UAS that created the
> >    early dialog has terminated it, and so the client MAY release
> >    resources and procedures associated with the early 
> dialog for which
> >    the 199 response is received.
> > 
> > * Section 4
> > 
> > Add this paragraph before the last paragraph:
> > 
> >    A UAC that receives a 199 response for an early dialog MUST NOT
> >    send any further requests on that dialog, except for a 
> PRACK if the
> >    199 response was sent reliably, and PRACKs for other reliable
> >    provisional responses that are to be acknowledged.
> > 
> > This is copied from section 7 (Backward Compatibility), but 
> since it 
> > is a constraint on UACs, it should also be stated in this section.
> > 
> > * Section 4
> > 
> >    The UAC MAY use additional information (for example the final
> >    response which triggered the 199 response) received in the 199
> >    response when initiating new sessions, if it is possible 
> to avoid the
> >    new session initiation request to be rejected.
> > 
> > What is the meaning of the final clause?
> > 
> > * Section 5
> > 
> > Add "early" as marked:
> > 
> >    When a UAS wants to terminate *an early* dialog it sends 
> a non-2xx
> >    SIP final response, as specified in [RFC3261].  In 
> addition, prior
> >    to sending the non-2xx SIP final response, the UAS MAY send a 199
> >    response to indicate that the dialog is being terminated.
> > 
> > This clarifies that 199 cannot be used to terminate a confirmed 
> > dialog.
> > 
> > (The reason a UAS may want to send a 199 is to provide this 
> > information even when the upstream proxy does not support 199.)
> > 
> > There is a subtlty to the second sentence:
> > 
> > Sending the 199 *before* the final response is not necessary due to 
> > the semantics of 199 (as it will probably reach the UAC 
> first anyway, 
> > and it would not matter if it did not), but rather to prevent the 
> > upstream proxy from generating a 199 based on the final 
> response, and 
> > then to transmit the UAS's 199 immediately after -- if the 
> UAS sends 
> > the 199 first, any upstream proxy will (most likely) see the 199 
> > before seeing the final response (which might induce the 
> proxy to send 
> > its own 199).
> > 
> > * Section 6
> > 
> >    When a forking proxy receives a non-2xx final response which
> >    terminates one or more early dialogs and the proxy does 
> not intend to
> >    forward the final response immediately (due to the rules for a
> >    forking proxy) it MUST send a 199 provisional response, for each
> >    associated early dialog that it can associate with the final
> >    response, upstream towards the sender of the associated request.
> > 
> > Clarify, and note that if the proxy has already passed a 199 for an 
> > early dialog, it need not generate one itself.  Also, I've reduced 
> > MUST to SHOULD, because (as the following NOTE explains), the proxy 
> > may not know of all early dialogs which are terminated by the final 
> > response.
> > 
> >    [...] it SHOULD initiate a 199 provisional response upstream
> >    towards the sender of the associated request, for each terminated
> >    early dialog, excepting those for which it has already 
> transmitted
> >    a 199 response.
> > 
> > Note that the "already transmitted 199 responses" may have 
> come from 
> > downstream proxies as well as downstream UASs, so this 
> consideration 
> > applies even if we do not allow UASs to send 199s.
> > 
> > * Section 7
> > 
> >    Since all SIP entities involved in a session setup do 
> not necessarily
> >    support the specific meaning of the 199 Early Dialog Terminated
> >    provisional response, the sender of the response MUST be 
> prepared to
> >    receive SIP requests associated with the dialog for which the 199
> >    response was sent.
> > 
> > This situation can also result due to race conditions.  Amend to:
> > 
> >    Because of race conditions and because the client may 
> not support the
> >    specific meaning of the 199 provisional response, the 
> sender of the
> >    response MUST be prepared to receive SIP requests associated with
> >    the dialog for which the 199 response was sent.
> > 
> > And add a qualifier to this sentence:
> > 
> >    If such request is received, and the receiver maintains state of
> >    the dialogs, the receiver MUST reply to such requests with a 481
> >    final response (unless another error response is more 
> appropriate).
> > 
> > * Section 7
> > 
> > Remove "..." from this sentence and add "even" as marked:
> > 
> >    The 199 Early Dialog Terminated response code MUST NOT replace a
> >    final response.  A final response MUST always be sent, 
> *even* after
> >    one or many 199 Early Dialog Terminated provisional 
> responses have
> >    been sent.
> > 
> > * Section 8
> > 
> > Clarify this sentence to:
> > 
> >    The 199 Early Dialog Terminated response code allows a SIP entity
> >    to indicate to upstream entities that a specific early dialog has
> >    been terminated, before a final response reaches the upstream
> >    entities.
> > 
> > Add the following:
> > 
> >    The early dialog that has been terminated is identified 
> by the Call-Id
> >    header and the to-tag value of the 199 response.
> > 
> >    Future standards may define optional additional 
> information carried
> >    in the headers and/or body of the response.
> > 
> > * Section 9
> > 
> >    A 199 Early Dialog Terminated provisional response MUST 
> NOT contain a
> >    new SDP offer/answer message body, but the sender of the 
> response MAY
> >    insert a copy of a previously sent offer/answer message body as
> >    otherwise allowed by the offer/answer rules for a provisional
> >    response.
> > 
> > This rule seems excessively strict, as it seems OK to send 
> a new SDP 
> > answer in a 199, even if the dialog will be immediately terminated.
> > Certainly, the UAC cannot tell the difference, as there could have 
> > been a previous but lost 1xx response carrying the same SDP.
> > 
> > The interaction of 199 and SDP probably needs further study...
> > 
> > * Section 11
> > 
> > Nit:  Use "reject" here rather than "rejects":
> > 
> >    UAS_2 and UAS_3 *reject* the INVITE by sending a 4xx 
> error response.
> > 
> > Here is a modification of the example, showing how a UAS can send a 
> > 199, and how the proxy handles it.  (You may or may not want to use
> > this.)  I've also added the ACK's for the 4xx's, which were 
> missing in 
> > the original.
> > 
> >    [...] UAS_2 and UAS_3 reject the INVITE by sending a 4xx error
> >    response.  UAS_3 sends a 199 with its 4xx.  When P1 receives the
> >    4xx response from UAS_2 it immediately sends a 199 Early Dialog
> >    Terminated associated with UAS_2's dialog towards the UAC.  P1
> >    transmits UAS_3's 199, and so does not initiate one itself for
> >    UAS_3's dialog when it receives UAS_3's 4xx response.
> > 
> >       UAC               P1                  UAS_2    UAS_3    UAS_4
> > 
> >        --- INVITE ------>
> >                          --- INVITE (leg 2) ->
> >                          --- INVITE (leg 3) ---------->
> >                          --- INVITE (leg 4) ------------------->
> >                          <-- 18x (leg 2) -----
> >        <-- 18x (leg 2) --
> >                          <-- 18x (leg 3) --------------
> >        <-- 18x (leg 3) --
> >                          <-- 18x (leg 4) -----------------------
> >        <-- 18x (leg 4) --
> >                          <-- 4xx (leg 2) -----
> >        <-- 199 (leg 2) --
> >                          --- ACK (leg 2) ---->
> >                          <-- 199 (leg 3) --------------
> >        <-- 199 (leg 3) --
> >                          <-- 4xx (leg 3) --------------
> >                          --- ACK (leg 3) ------------->
> >                          <-- 200 (leg 4) -----------------------
> >        <-- 200 (leg 4) --
> >        --- ACK (leg 4) ->
> >                          --- ACK (leg 4) ---------------------->
> > 
> > Dale
> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use 
> > sip-implementors@cs.columbia.edu for questions on current sip Use 
> > sipping@ietf.org for new developments on the application of sip
> > 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use 
> sip-implementors@cs.columbia.edu for questions on current sip 
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
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From: Paul Kyzivat <pkyzivat@cisco.com>
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Francois Audet wrote:
> I can think of one scenario...
> 
> If the UAS in question is actually some sort of B2BUA. 

OK. That is a good one.

That is certainly a reason not to forbid it.

And in that case we would expect the B2BUA to follow the rules for 
proxies about handling and generating 199.

	Thanks,
	Paul

>> -----Original Message-----
>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
>> Behalf Of Paul Kyzivat
>> Sent: Friday, June 13, 2008 07:07
>> To: Dale.Worley@comcast.net
>> Cc: sip@ietf.org
>> Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
>>
>> This all sounds like good stuff.
>>
>> However I am still scratching my head about the UAS sending a 
>> 199. While I see no reason to forbid it I'm trying to find a 
>> situation when it would be helpful. The only case I have been 
>> able to think of is:
>>
>>    There is a forking proxy upstream of the UAS, but it doesn't
>>    support 199. Yet it may delay the final response while it
>>    tries other forks. So the UAS sending the 199 makes up for
>>    the inadequacy of the forking proxy.
>>
>> The trouble with this is that it is inefficient in many 
>> cases. If each proxy that forks also supports the sending of 
>> 199, and the UAS doesn't send one, then we avoid sending the 
>> message except in cases where it is helpful. If the UAS 
>> always sends it we have increased the message count by one 
>> for all failing calls. And the majority of failing calls get 
>> no benefit from a 199.
>>
>> This is one of those cases that could probably benefit from 
>> simulation to judge the merits. My gut feel is that the costs 
>> of the UAS sending the 199 outweigh the benefits. Barring 
>> some analysis to the contrary I think I would say that a UAS 
>> SHOULD NOT send a 199. Or perhaps MAY send, but without any 
>> guidance of when.
>>
>> 	Paul
>>
>> Dale.Worley@comcast.net wrote:
>>> Here are some comments on this I-D.  I don't think that any of them 
>>> modify our understanding of the proposal.  I guess this is 
>> very terse; 
>>> please use these suggestions as you see fit.
>>>
>>> * Abstract
>>>
>>> Clarify:
>>>
>>>    This specification defines a new SIP response code, 199 Early
>>>    Dialog Terminated, which a SIP entity can use to 
>> indicate upstream
>>>    that an early dialog has been terminated.  The response 
>> code can be
>>>    used by a SIP entity (a SIP proxy or UAS), to inform the UAC that
>>>    an early dialog has been terminated.  As the 199 response is a
>>>    provisional response, often the UAC will receive it before
>>>    receiving a final response to its request.
>>>
>>> * Section 1
>>>
>>>    When a forking proxy receives a non-2xx final response, 
>> it does not
>>>    always immediately forward the response upstream towards 
>> the sender
>>>    of the associated request.  Instead, the forking proxy 
>> "stores" it
>>>    and waits for further final responses from remote 
>> destinations where
>>>    the forked request was forwarded.
>>>
>>> The text does not allow for serial forking, only parallel forking, 
>>> which might be confusing to people who are more aware of serial 
>>> forking.  So amend the last sentence to:
>>>
>>>    Instead, the forking proxy "stores" it and waits for 
>> further final
>>>    responses from remote destinations where the forked request was
>>>    forwarded, and possibly forwards the request to additional
>>>    destinations.
>>>
>>> Also, Avoid using "..." except when indicating that the 
>> word is used 
>>> in a non-standard way or that one is discussing the word itself:
>>>
>>>    Instead, the forking proxy "stores" it [...]
>>>
>>> should be:
>>>
>>>    Instead, the forking proxy stores it [...]
>>>
>>> * Section 3.
>>>
>>>    REQ 1: It must be possible to indicate to the UAC that an early
>>>    dialog has been terminated before a final response is sent.
>>>
>>> Oddly, though everyone thinks they understand this sentence, it 
>>> doesn't really say what we want.  To really ensure that the UAC 
>>> understands the early dialog has been terminated before the 
>> UAS sends 
>>> a final response, the UAS would have to send 199 *and get the PRACK 
>>> saying that it was received* before sending a final response.  This 
>>> may well be worse than the present state of affairs.
>>>
>>> What we really mean is:
>>>
>>>    REQ 1: It must be possible for a UAS or proxy to send an 
>> indication
>>>    to the UAC that an early dialog has been terminated, which
>>>    indication will not be delayed waiting for a final response from
>>>    any UAS.
>>>
>>> * Section 4
>>>
>>>    When a client receives a 199 Early Dialog Terminated provisional
>>>    response it MAY release resources and procedures 
>> associated with the
>>>    early dialog the 199 response is received on.
>>>
>>> Amend to clarify the meaning of 199 independently of the 
>> consequences 
>>> of that meaning:
>>>
>>>    When a client receives a 199 Early Dialog Terminated provisional
>>>    response, it has positive knowledge that the UAS that created the
>>>    early dialog has terminated it, and so the client MAY release
>>>    resources and procedures associated with the early 
>> dialog for which
>>>    the 199 response is received.
>>>
>>> * Section 4
>>>
>>> Add this paragraph before the last paragraph:
>>>
>>>    A UAC that receives a 199 response for an early dialog MUST NOT
>>>    send any further requests on that dialog, except for a 
>> PRACK if the
>>>    199 response was sent reliably, and PRACKs for other reliable
>>>    provisional responses that are to be acknowledged.
>>>
>>> This is copied from section 7 (Backward Compatibility), but 
>> since it 
>>> is a constraint on UACs, it should also be stated in this section.
>>>
>>> * Section 4
>>>
>>>    The UAC MAY use additional information (for example the final
>>>    response which triggered the 199 response) received in the 199
>>>    response when initiating new sessions, if it is possible 
>> to avoid the
>>>    new session initiation request to be rejected.
>>>
>>> What is the meaning of the final clause?
>>>
>>> * Section 5
>>>
>>> Add "early" as marked:
>>>
>>>    When a UAS wants to terminate *an early* dialog it sends 
>> a non-2xx
>>>    SIP final response, as specified in [RFC3261].  In 
>> addition, prior
>>>    to sending the non-2xx SIP final response, the UAS MAY send a 199
>>>    response to indicate that the dialog is being terminated.
>>>
>>> This clarifies that 199 cannot be used to terminate a confirmed 
>>> dialog.
>>>
>>> (The reason a UAS may want to send a 199 is to provide this 
>>> information even when the upstream proxy does not support 199.)
>>>
>>> There is a subtlty to the second sentence:
>>>
>>> Sending the 199 *before* the final response is not necessary due to 
>>> the semantics of 199 (as it will probably reach the UAC 
>> first anyway, 
>>> and it would not matter if it did not), but rather to prevent the 
>>> upstream proxy from generating a 199 based on the final 
>> response, and 
>>> then to transmit the UAS's 199 immediately after -- if the 
>> UAS sends 
>>> the 199 first, any upstream proxy will (most likely) see the 199 
>>> before seeing the final response (which might induce the 
>> proxy to send 
>>> its own 199).
>>>
>>> * Section 6
>>>
>>>    When a forking proxy receives a non-2xx final response which
>>>    terminates one or more early dialogs and the proxy does 
>> not intend to
>>>    forward the final response immediately (due to the rules for a
>>>    forking proxy) it MUST send a 199 provisional response, for each
>>>    associated early dialog that it can associate with the final
>>>    response, upstream towards the sender of the associated request.
>>>
>>> Clarify, and note that if the proxy has already passed a 199 for an 
>>> early dialog, it need not generate one itself.  Also, I've reduced 
>>> MUST to SHOULD, because (as the following NOTE explains), the proxy 
>>> may not know of all early dialogs which are terminated by the final 
>>> response.
>>>
>>>    [...] it SHOULD initiate a 199 provisional response upstream
>>>    towards the sender of the associated request, for each terminated
>>>    early dialog, excepting those for which it has already 
>> transmitted
>>>    a 199 response.
>>>
>>> Note that the "already transmitted 199 responses" may have 
>> come from 
>>> downstream proxies as well as downstream UASs, so this 
>> consideration 
>>> applies even if we do not allow UASs to send 199s.
>>>
>>> * Section 7
>>>
>>>    Since all SIP entities involved in a session setup do 
>> not necessarily
>>>    support the specific meaning of the 199 Early Dialog Terminated
>>>    provisional response, the sender of the response MUST be 
>> prepared to
>>>    receive SIP requests associated with the dialog for which the 199
>>>    response was sent.
>>>
>>> This situation can also result due to race conditions.  Amend to:
>>>
>>>    Because of race conditions and because the client may 
>> not support the
>>>    specific meaning of the 199 provisional response, the 
>> sender of the
>>>    response MUST be prepared to receive SIP requests associated with
>>>    the dialog for which the 199 response was sent.
>>>
>>> And add a qualifier to this sentence:
>>>
>>>    If such request is received, and the receiver maintains state of
>>>    the dialogs, the receiver MUST reply to such requests with a 481
>>>    final response (unless another error response is more 
>> appropriate).
>>> * Section 7
>>>
>>> Remove "..." from this sentence and add "even" as marked:
>>>
>>>    The 199 Early Dialog Terminated response code MUST NOT replace a
>>>    final response.  A final response MUST always be sent, 
>> *even* after
>>>    one or many 199 Early Dialog Terminated provisional 
>> responses have
>>>    been sent.
>>>
>>> * Section 8
>>>
>>> Clarify this sentence to:
>>>
>>>    The 199 Early Dialog Terminated response code allows a SIP entity
>>>    to indicate to upstream entities that a specific early dialog has
>>>    been terminated, before a final response reaches the upstream
>>>    entities.
>>>
>>> Add the following:
>>>
>>>    The early dialog that has been terminated is identified 
>> by the Call-Id
>>>    header and the to-tag value of the 199 response.
>>>
>>>    Future standards may define optional additional 
>> information carried
>>>    in the headers and/or body of the response.
>>>
>>> * Section 9
>>>
>>>    A 199 Early Dialog Terminated provisional response MUST 
>> NOT contain a
>>>    new SDP offer/answer message body, but the sender of the 
>> response MAY
>>>    insert a copy of a previously sent offer/answer message body as
>>>    otherwise allowed by the offer/answer rules for a provisional
>>>    response.
>>>
>>> This rule seems excessively strict, as it seems OK to send 
>> a new SDP 
>>> answer in a 199, even if the dialog will be immediately terminated.
>>> Certainly, the UAC cannot tell the difference, as there could have 
>>> been a previous but lost 1xx response carrying the same SDP.
>>>
>>> The interaction of 199 and SDP probably needs further study...
>>>
>>> * Section 11
>>>
>>> Nit:  Use "reject" here rather than "rejects":
>>>
>>>    UAS_2 and UAS_3 *reject* the INVITE by sending a 4xx 
>> error response.
>>> Here is a modification of the example, showing how a UAS can send a 
>>> 199, and how the proxy handles it.  (You may or may not want to use
>>> this.)  I've also added the ACK's for the 4xx's, which were 
>> missing in 
>>> the original.
>>>
>>>    [...] UAS_2 and UAS_3 reject the INVITE by sending a 4xx error
>>>    response.  UAS_3 sends a 199 with its 4xx.  When P1 receives the
>>>    4xx response from UAS_2 it immediately sends a 199 Early Dialog
>>>    Terminated associated with UAS_2's dialog towards the UAC.  P1
>>>    transmits UAS_3's 199, and so does not initiate one itself for
>>>    UAS_3's dialog when it receives UAS_3's 4xx response.
>>>
>>>       UAC               P1                  UAS_2    UAS_3    UAS_4
>>>
>>>        --- INVITE ------>
>>>                          --- INVITE (leg 2) ->
>>>                          --- INVITE (leg 3) ---------->
>>>                          --- INVITE (leg 4) ------------------->
>>>                          <-- 18x (leg 2) -----
>>>        <-- 18x (leg 2) --
>>>                          <-- 18x (leg 3) --------------
>>>        <-- 18x (leg 3) --
>>>                          <-- 18x (leg 4) -----------------------
>>>        <-- 18x (leg 4) --
>>>                          <-- 4xx (leg 2) -----
>>>        <-- 199 (leg 2) --
>>>                          --- ACK (leg 2) ---->
>>>                          <-- 199 (leg 3) --------------
>>>        <-- 199 (leg 3) --
>>>                          <-- 4xx (leg 3) --------------
>>>                          --- ACK (leg 3) ------------->
>>>                          <-- 200 (leg 4) -----------------------
>>>        <-- 200 (leg 4) --
>>>        --- ACK (leg 4) ->
>>>                          --- ACK (leg 4) ---------------------->
>>>
>>> Dale
>>> _______________________________________________
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>>> This list is for NEW development of the core SIP Protocol Use 
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>>> sipping@ietf.org for new developments on the application of sip
>>>
>> _______________________________________________
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>>
> 
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From: "Francois Audet" <audet@nortel.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
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Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
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Yeah, exactly. 


> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
> Sent: Friday, June 13, 2008 12:11
> To: Audet, Francois (SC100:3055)
> Cc: Dale.Worley@comcast.net; sip@ietf.org
> Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
> 
> 
> 
> Francois Audet wrote:
> > I can think of one scenario...
> > 
> > If the UAS in question is actually some sort of B2BUA. 
> 
> OK. That is a good one.
> 
> That is certainly a reason not to forbid it.
> 
> And in that case we would expect the B2BUA to follow the 
> rules for proxies about handling and generating 199.
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Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
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So the text certainly needs to permit this case.

But I think some care is needed about a UAS that is really at the end of 
the end of the stream (downstream, where the packets soak into the 
ground.) Do we want to allow that UAS to send 199 responses? If so, what 
if any guidelines should be given?

Seems to me we need a pretty strong SHOULD NOT here, with the B2BUA 
being the known exception.

	Paul

Francois Audet wrote:
> Yeah, exactly. 
> 
> 
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
>> Sent: Friday, June 13, 2008 12:11
>> To: Audet, Francois (SC100:3055)
>> Cc: Dale.Worley@comcast.net; sip@ietf.org
>> Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
>>
>>
>>
>> Francois Audet wrote:
>>> I can think of one scenario...
>>>
>>> If the UAS in question is actually some sort of B2BUA. 
>> OK. That is a good one.
>>
>> That is certainly a reason not to forbid it.
>>
>> And in that case we would expect the B2BUA to follow the 
>> rules for proxies about handling and generating 199.
> 
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Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
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   From: Paul Kyzivat <pkyzivat@cisco.com>

   However I am still scratching my head about the UAS sending a 199. While 
   I see no reason to forbid it I'm trying to find a situation when it 
   would be helpful. The only case I have been able to think of is:

I admit I hadn't considered whether a UAS sending 199 is useful.

But in giving the rules for proxies, we must take care of the case
where a downstream proxy sends a 199.  This is where we get the rule
that if a proxy propagates a 199 for an early dialog, then it should
not send a 199 for that early dialog itself.  Once we've set that rule
for proxies, then a UAS can also send a 199 without breaking anything.

But it does look like there is no gain from sending a 199 unless the
corresponding final response will not be sent immediately.

Dale
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From sip-bounces@ietf.org  Fri Jun 13 15:17:26 2008
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From: "Francois Audet" <audet@nortel.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
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Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
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Yeah, I think we should say that this feature is intended to be used by Proxies and
not by UAS. The reason it is allowed for UAS is for B2BUA who often act in many ways
like a proxy. 

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
> Sent: Friday, June 13, 2008 14:04
> To: Audet, Francois (SC100:3055)
> Cc: Dale.Worley@comcast.net; sip@ietf.org
> Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
> 
> So the text certainly needs to permit this case.
> 
> But I think some care is needed about a UAS that is really at 
> the end of the end of the stream (downstream, where the 
> packets soak into the
> ground.) Do we want to allow that UAS to send 199 responses? 
> If so, what if any guidelines should be given?
> 
> Seems to me we need a pretty strong SHOULD NOT here, with the 
> B2BUA being the known exception.
> 
> 	Paul
> 
> Francois Audet wrote:
> > Yeah, exactly. 
> > 
> > 
> >> -----Original Message-----
> >> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> >> Sent: Friday, June 13, 2008 12:11
> >> To: Audet, Francois (SC100:3055)
> >> Cc: Dale.Worley@comcast.net; sip@ietf.org
> >> Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
> >>
> >>
> >>
> >> Francois Audet wrote:
> >>> I can think of one scenario...
> >>>
> >>> If the UAS in question is actually some sort of B2BUA. 
> >> OK. That is a good one.
> >>
> >> That is certainly a reason not to forbid it.
> >>
> >> And in that case we would expect the B2BUA to follow the rules for 
> >> proxies about handling and generating 199.
> > 
> 
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From: jh@tutpro.com (Juha Heinanen)
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Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
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Paul Kyzivat writes:

 > Seems to me we need a pretty strong SHOULD NOT here, with the B2BUA 
 > being the known exception.

i don't agree excepting B2BUA for anything as long as b2bua has been
officially added and defined as a new sip related entity in addition to
UAs, registrars and proxies.  once that is done, then every sip
specification has to state if/when ua procedures also apply to these
B2BUAs.

-- juha
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Hi Juha,
 
Chapter 6 of RFC3261 DOES define Back-to-Back User Agent.
 
 Back-to-Back User Agent: A back-to-back user agent (B2BUA) is a
         logical entity that receives a request and processes it as a
         user agent server (UAS).  In order to determine how the request
         should be answered, it acts as a user agent client (UAC) and
         generates requests.  Unlike a proxy server, it maintains dialog
         state and must participate in all requests sent on the dialogs
         it has established.  Since it is a concatenation of a UAC and
         UAS, no explicit definitions are needed for its behavior.

We could of course say something like:
 
"UAS, when part of a B2BUA"
 
Regards,
 
Christer
 

________________________________

From: sip-bounces@ietf.org on behalf of Juha Heinanen
Sent: Sat 14/06/2008 08:59
To: Paul Kyzivat
Cc: sip@ietf.org; Francois Audet
Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00



Paul Kyzivat writes:

 > Seems to me we need a pretty strong SHOULD NOT here, with the B2BUA
 > being the known exception.

i don't agree excepting B2BUA for anything as long as b2bua has been
officially added and defined as a new sip related entity in addition to
UAs, registrars and proxies.  once that is done, then every sip
specification has to state if/when ua procedures also apply to these
B2BUAs.

-- juha
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Hi,
 
I think Paul's suggestion sounds good. I think it's also alligned with what Francois suggested earlier, to say "MUST include response code".
 
Regards,
 
Christer

________________________________

From: sip-bounces@ietf.org on behalf of Paul Kyzivat
Sent: Fri 13/06/2008 16:39
To: Francois Audet
Cc: SIP IETF
Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt



My first thought was to agree with Dale, and in fact I already suggested
use of sipfrag. But I already have experience from the refer NOTIFY that
sipfrag info is difficult to take advantage of if you have no guarantee
of what will be included.

IIRC this 199 work was explicitly not expected to solve the full HERFP
problem. Yet if it could serve for that it would be a good thing. Its
just that thrashing out everything that is needed will take a bunch of time.

Also, sending back the whole message will make more data to send in
cases where it often isn't useful.

We could allow a body containing *either* message/sip or
message/sipfrag, and make them both conditional on presence of a
corresponding Accept header in the request. If there was no Accept
header for either then no body would be included. If there was an accept
for both, message/sip would be be preferred. When a sipfrag is returned
in a 199, we could require that it include at least the header line, and
the rest would be at the discretion of the proxy.

        Paul

Francois Audet wrote:
> And do you think we should specify what are the minimum headers thatshould
> be carried?
>
> What about MIME bodies?
>
> There might be cases where they are really important, no?
>
> I'm a little worried that the proxy may remove important information.
> Especially when we extend SIP.
>
> Maybe instead we should list the removable headers (e.g., all the ones that
> are identical to their counterpart in the 199 itself), and then explain
> "don't remove what you don't understand"?
>
> Opinions?
>
>
> On Jun12 2008 20:05 , "Dale.Worley@comcast.net" <Dale.Worley@comcast.net>
> wrote:
>
>>    From: "Francois Audet" <audet@nortel.com>
>>
>>    I really think that this draft should make the inclusion of the original
>>    non-2XX final response in the 199 MANDATORY by the forking proxy. This
>> would
>>    be in a message/sip body as described in sectin 23.4/RFC 3261.
>>
>> (This is outside the scope of the I-D itself.)
>>
>> If we envision having the 199 carry information from the associated
>> failure response, I suggest we use a message/sipfrag body excerpted
>> from the failure response, rather than copying headers from the
>> failure response into the 199's headers, or copying the entire failure
>> response as the body of the 199.
>>
>> The syntax of sipfrag is unambiguous but extremely flexible (RFC
>> 3420):
>>
>>            sipfrag = [ start-line ]
>>                      *message-header
>>                      [ CRLF [ message-body ] ]
>>
>> Essentially, it is any subset of the headers of the response,
>> optionally followed by the body.  This subsetting capability allows
>> small responses (e.g., omit the Via's, Call-Id, To, From,
>> Record-Route's, etc.), as well as making topology-hiding
>> straightforward.
>>
>> It's also inherently extensible, in that if a proxy/UAS decides to add
>> more headers to the sipfrag, the UAC knows how to identify and/or
>> ignore them.
>>
>> It's also terse, in that carrying any header from the response in a
>> message/sipfrag body is no longer than carrying it as a header of the
>> 199 response.
>>
>> Dale
>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>
> _______________________________________________
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>
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Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
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Christer Holmberg writes:

 >  Back-to-Back User Agent: A back-to-back user agent (B2BUA) is a
 >          logical entity that receives a request and processes it as a
 >          user agent server (UAS).  In order to determine how the request
 >          should be answered, it acts as a user agent client (UAC) and
 >          generates requests.  Unlike a proxy server, it maintains dialog
 >          state and must participate in all requests sent on the dialogs
 >          it has established.  
 > 
 > We could of course say something like:
 >  
 > "UAS, when part of a B2BUA"

according to the B2BUA definition:

	Since it is a concatenation of a UAC and
      UAS, no explicit definitions are needed for its behavior.

this means that there is no need to say anything special about 199
either.  b2bus in terms of protocol specification, is just an uas like
any other.

-- juha
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Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
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Well, the specification could be abstracted to remove the mention of the 
B2BUA while still having the same effect. It would have to say something 
about a UAS should send a 199 if it knows the dialog will be terminated 
before it can send the final response. But I think we would still need 
to include an example of a B2BUA in order for anybody to have a clue 
what we were talking about.

	Paul

Juha Heinanen wrote:
> Christer Holmberg writes:
> 
>  >  Back-to-Back User Agent: A back-to-back user agent (B2BUA) is a
>  >          logical entity that receives a request and processes it as a
>  >          user agent server (UAS).  In order to determine how the request
>  >          should be answered, it acts as a user agent client (UAC) and
>  >          generates requests.  Unlike a proxy server, it maintains dialog
>  >          state and must participate in all requests sent on the dialogs
>  >          it has established.  
>  > 
>  > We could of course say something like:
>  >  
>  > "UAS, when part of a B2BUA"
> 
> according to the B2BUA definition:
> 
> 	Since it is a concatenation of a UAC and
>       UAS, no explicit definitions are needed for its behavior.
> 
> this means that there is no need to say anything special about 199
> either.  b2bus in terms of protocol specification, is just an uas like
> any other.
> 
> -- juha
> 
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Thread-Topic: [Sip] Draft submission: draft-ietf-sip-199-00
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From: "Attila Sipos" <attila.sipos@vegastream.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>, <Dale.Worley@comcast.net>
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Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
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>>However I am still scratching my head about the UAS sending a 199. While=
=20
>>I see no reason to forbid it I'm trying to find a situation when it=20
>>would be helpful.=20

It may or may not be helpful but as you say, you can
see no reason to forbid it.

Why restrict it if the restriction isn't needed?

As someone has pointed out, a B2BUA scenario might benefit
from 199.  In fact, any kind of SIP gateway might benefit.

> This is one of those cases that could probably benefit from
> simulation to judge the merits. My gut feel is that the costs
> of the UAS sending the 199 outweigh the benefits.=20

If the UAS wants to send it, that is a decision for
them - not something that needs to be prevented.

SIP forking, in the technical sense of sending a
response with a different to-tag, is always legal.
So, any UA which has used forking
should also be able to use 199.

I don't think we need to include any specific text about UAS
except to say that a UAS can use it if the UAS has created
the forking.

Regards,

Attila



-----Original Message-----
From: sip-bounces@ietf.org on behalf of Paul Kyzivat
Sent: Fri 13/06/2008 15:07
To: Dale.Worley@comcast.net
Cc: sip@ietf.org
Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
=20
This all sounds like good stuff.

However I am still scratching my head about the UAS sending a 199. While=20
I see no reason to forbid it I'm trying to find a situation when it=20
would be helpful. The only case I have been able to think of is:

   There is a forking proxy upstream of the UAS, but it doesn't
   support 199. Yet it may delay the final response while it
   tries other forks. So the UAS sending the 199 makes up for
   the inadequacy of the forking proxy.

The trouble with this is that it is inefficient in many cases. If each=20
proxy that forks also supports the sending of 199, and the UAS doesn't=20
send one, then we avoid sending the message except in cases where it is=20
helpful. If the UAS always sends it we have increased the message count=20
by one for all failing calls. And the majority of failing calls get no=20
benefit from a 199.

This is one of those cases that could probably benefit from simulation=20
to judge the merits. My gut feel is that the costs of the UAS sending=20
the 199 outweigh the benefits. Barring some analysis to the contrary I=20
think I would say that a UAS SHOULD NOT send a 199. Or perhaps MAY send,=20
but without any guidance of when.

	Paul

Dale.Worley@comcast.net wrote:
> Here are some comments on this I-D.  I don't think that any of them
> modify our understanding of the proposal.  I guess this is very terse;
> please use these suggestions as you see fit.
>=20
> * Abstract
>=20
> Clarify:
>=20
>    This specification defines a new SIP response code, 199 Early
>    Dialog Terminated, which a SIP entity can use to indicate upstream
>    that an early dialog has been terminated.  The response code can be
>    used by a SIP entity (a SIP proxy or UAS), to inform the UAC that
>    an early dialog has been terminated.  As the 199 response is a
>    provisional response, often the UAC will receive it before
>    receiving a final response to its request.
>=20
> * Section 1
>=20
>    When a forking proxy receives a non-2xx final response, it does not
>    always immediately forward the response upstream towards the sender
>    of the associated request.  Instead, the forking proxy "stores" it
>    and waits for further final responses from remote destinations where
>    the forked request was forwarded.
>=20
> The text does not allow for serial forking, only parallel forking,
> which might be confusing to people who are more aware of serial
> forking.  So amend the last sentence to:
>=20
>    Instead, the forking proxy "stores" it and waits for further final
>    responses from remote destinations where the forked request was
>    forwarded, and possibly forwards the request to additional
>    destinations.
>=20
> Also, Avoid using "..." except when indicating that the word is used
> in a non-standard way or that one is discussing the word itself:
>=20
>    Instead, the forking proxy "stores" it [...]
>=20
> should be:
>=20
>    Instead, the forking proxy stores it [...]
>=20
> * Section 3.
>=20
>    REQ 1: It must be possible to indicate to the UAC that an early
>    dialog has been terminated before a final response is sent.
>=20
> Oddly, though everyone thinks they understand this sentence, it
> doesn't really say what we want.  To really ensure that the UAC
> understands the early dialog has been terminated before the UAS sends
> a final response, the UAS would have to send 199 *and get the PRACK
> saying that it was received* before sending a final response.  This may
> well be worse than the present state of affairs.
>=20
> What we really mean is:
>=20
>    REQ 1: It must be possible for a UAS or proxy to send an indication
>    to the UAC that an early dialog has been terminated, which
>    indication will not be delayed waiting for a final response from
>    any UAS.
>=20
> * Section 4
>=20
>    When a client receives a 199 Early Dialog Terminated provisional
>    response it MAY release resources and procedures associated with the
>    early dialog the 199 response is received on.
>=20
> Amend to clarify the meaning of 199 independently of the consequences
> of that meaning:
>=20
>    When a client receives a 199 Early Dialog Terminated provisional
>    response, it has positive knowledge that the UAS that created the
>    early dialog has terminated it, and so the client MAY release
>    resources and procedures associated with the early dialog for which
>    the 199 response is received.
>=20
> * Section 4
>=20
> Add this paragraph before the last paragraph:
>=20
>    A UAC that receives a 199 response for an early dialog MUST NOT
>    send any further requests on that dialog, except for a PRACK if the
>    199 response was sent reliably, and PRACKs for other reliable
>    provisional responses that are to be acknowledged.
>=20
> This is copied from section 7 (Backward Compatibility), but since it
> is a constraint on UACs, it should also be stated in this section.
>=20
> * Section 4
>=20
>    The UAC MAY use additional information (for example the final
>    response which triggered the 199 response) received in the 199
>    response when initiating new sessions, if it is possible to avoid the
>    new session initiation request to be rejected.
>=20
> What is the meaning of the final clause?
>=20
> * Section 5
>=20
> Add "early" as marked:
>=20
>    When a UAS wants to terminate *an early* dialog it sends a non-2xx
>    SIP final response, as specified in [RFC3261].  In addition, prior
>    to sending the non-2xx SIP final response, the UAS MAY send a 199
>    response to indicate that the dialog is being terminated.
>=20
> This clarifies that 199 cannot be used to terminate a confirmed
> dialog.
>=20
> (The reason a UAS may want to send a 199 is to provide this
> information even when the upstream proxy does not support 199.)
>=20
> There is a subtlty to the second sentence:
>=20
> Sending the 199 *before* the final response is not necessary due to
> the semantics of 199 (as it will probably reach the UAC first anyway,
> and it would not matter if it did not), but rather to prevent the
> upstream proxy from generating a 199 based on the final response, and
> then to transmit the UAS's 199 immediately after -- if the UAS sends
> the 199 first, any upstream proxy will (most likely) see the 199
> before seeing the final response (which might induce the proxy to send
> its own 199).
>=20
> * Section 6
>=20
>    When a forking proxy receives a non-2xx final response which
>    terminates one or more early dialogs and the proxy does not intend to
>    forward the final response immediately (due to the rules for a
>    forking proxy) it MUST send a 199 provisional response, for each
>    associated early dialog that it can associate with the final
>    response, upstream towards the sender of the associated request.
>=20
> Clarify, and note that if the proxy has already passed a 199 for an
> early dialog, it need not generate one itself.  Also, I've reduced
> MUST to SHOULD, because (as the following NOTE explains), the proxy
> may not know of all early dialogs which are terminated by the final
> response.
>=20
>    [...] it SHOULD initiate a 199 provisional response upstream
>    towards the sender of the associated request, for each terminated
>    early dialog, excepting those for which it has already transmitted
>    a 199 response.
>=20
> Note that the "already transmitted 199 responses" may have come from
> downstream proxies as well as downstream UASs, so this consideration
> applies even if we do not allow UASs to send 199s.
>=20
> * Section 7
>=20
>    Since all SIP entities involved in a session setup do not necessarily
>    support the specific meaning of the 199 Early Dialog Terminated
>    provisional response, the sender of the response MUST be prepared to
>    receive SIP requests associated with the dialog for which the 199
>    response was sent.
>=20
> This situation can also result due to race conditions.  Amend to:
>=20
>    Because of race conditions and because the client may not support the
>    specific meaning of the 199 provisional response, the sender of the
>    response MUST be prepared to receive SIP requests associated with
>    the dialog for which the 199 response was sent.
>=20
> And add a qualifier to this sentence:
>=20
>    If such request is received, and the receiver maintains state of
>    the dialogs, the receiver MUST reply to such requests with a 481
>    final response (unless another error response is more appropriate).
>=20
> * Section 7
>=20
> Remove "..." from this sentence and add "even" as marked:
>=20
>    The 199 Early Dialog Terminated response code MUST NOT replace a
>    final response.  A final response MUST always be sent, *even* after
>    one or many 199 Early Dialog Terminated provisional responses have
>    been sent.
>=20
> * Section 8
>=20
> Clarify this sentence to:
>=20
>    The 199 Early Dialog Terminated response code allows a SIP entity
>    to indicate to upstream entities that a specific early dialog has
>    been terminated, before a final response reaches the upstream
>    entities.
>=20
> Add the following:
>=20
>    The early dialog that has been terminated is identified by the Call-Id
>    header and the to-tag value of the 199 response.
>=20
>    Future standards may define optional additional information carried
>    in the headers and/or body of the response.
>=20
> * Section 9
>=20
>    A 199 Early Dialog Terminated provisional response MUST NOT contain a
>    new SDP offer/answer message body, but the sender of the response MAY
>    insert a copy of a previously sent offer/answer message body as
>    otherwise allowed by the offer/answer rules for a provisional
>    response.
>=20
> This rule seems excessively strict, as it seems OK to send a new SDP
> answer in a 199, even if the dialog will be immediately terminated.
> Certainly, the UAC cannot tell the difference, as there could have
> been a previous but lost 1xx response carrying the same SDP.
>=20
> The interaction of 199 and SDP probably needs further study...
>=20
> * Section 11
>=20
> Nit:  Use "reject" here rather than "rejects":
>=20
>    UAS_2 and UAS_3 *reject* the INVITE by sending a 4xx error response.
>=20
> Here is a modification of the example, showing how a UAS can send a
> 199, and how the proxy handles it.  (You may or may not want to use
> this.)  I've also added the ACK's for the 4xx's, which were missing in
> the original.
>=20
>    [...] UAS_2 and UAS_3 reject the INVITE by sending a 4xx error
>    response.  UAS_3 sends a 199 with its 4xx.  When P1 receives the
>    4xx response from UAS_2 it immediately sends a 199 Early Dialog
>    Terminated associated with UAS_2's dialog towards the UAC.  P1
>    transmits UAS_3's 199, and so does not initiate one itself for
>    UAS_3's dialog when it receives UAS_3's 4xx response.
>=20
>       UAC               P1                  UAS_2    UAS_3    UAS_4
>=20
>        --- INVITE ------>
>                          --- INVITE (leg 2) ->
>                          --- INVITE (leg 3) ---------->
>                          --- INVITE (leg 4) ------------------->
>                          <-- 18x (leg 2) -----
>        <-- 18x (leg 2) --
>                          <-- 18x (leg 3) --------------
>        <-- 18x (leg 3) --
>                          <-- 18x (leg 4) -----------------------
>        <-- 18x (leg 4) --
>                          <-- 4xx (leg 2) -----
>        <-- 199 (leg 2) --
>                          --- ACK (leg 2) ---->
>                          <-- 199 (leg 3) --------------
>        <-- 199 (leg 3) --
>                          <-- 4xx (leg 3) --------------
>                          --- ACK (leg 3) ------------->
>                          <-- 200 (leg 4) -----------------------
>        <-- 200 (leg 4) --
>        --- ACK (leg 4) ->
>                          --- ACK (leg 4) ---------------------->
>=20
> Dale
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


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<TITLE>RE: [Sip] Draft submission: draft-ietf-sip-199-00</TITLE>
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<P><FONT SIZE=3D2>&gt;&gt;However I am still scratching my head about the U=
AS sending a 199. While<BR>
&gt;&gt;I see no reason to forbid it I'm trying to find a situation when it=
<BR>
&gt;&gt;would be helpful.<BR>
<BR>
It may or may not be helpful but as you say, you can<BR>
see no reason to forbid it.<BR>
<BR>
Why restrict it if the restriction isn't needed?<BR>
<BR>
As someone has pointed out, a B2BUA scenario might benefit<BR>
from 199.&nbsp; In fact, any kind of SIP gateway might benefit.<BR>
<BR>
&gt; This is one of those cases that could probably benefit from<BR>
&gt; simulation to judge the merits. My gut feel is that the costs<BR>
&gt; of the UAS sending the 199 outweigh the benefits.<BR>
<BR>
If the UAS wants to send it, that is a decision for<BR>
them - not something that needs to be prevented.<BR>
<BR>
SIP forking, in the technical sense of sending a<BR>
response with a different to-tag, is always legal.<BR>
So, any UA which has used forking<BR>
should also be able to use 199.<BR>
<BR>
I don't think we need to include any specific text about UAS<BR>
except to say that a UAS can use it if the UAS has created<BR>
the forking.<BR>
<BR>
Regards,<BR>
<BR>
Attila<BR>
<BR>
<BR>
<BR>
-----Original Message-----<BR>
From: sip-bounces@ietf.org on behalf of Paul Kyzivat<BR>
Sent: Fri 13/06/2008 15:07<BR>
To: Dale.Worley@comcast.net<BR>
Cc: sip@ietf.org<BR>
Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00<BR>
<BR>
This all sounds like good stuff.<BR>
<BR>
However I am still scratching my head about the UAS sending a 199. While<BR>
I see no reason to forbid it I'm trying to find a situation when it<BR>
would be helpful. The only case I have been able to think of is:<BR>
<BR>
&nbsp;&nbsp; There is a forking proxy upstream of the UAS, but it doesn't<B=
R>
&nbsp;&nbsp; support 199. Yet it may delay the final response while it<BR>
&nbsp;&nbsp; tries other forks. So the UAS sending the 199 makes up for<BR>
&nbsp;&nbsp; the inadequacy of the forking proxy.<BR>
<BR>
The trouble with this is that it is inefficient in many cases. If each<BR>
proxy that forks also supports the sending of 199, and the UAS doesn't<BR>
send one, then we avoid sending the message except in cases where it is<BR>
helpful. If the UAS always sends it we have increased the message count<BR>
by one for all failing calls. And the majority of failing calls get no<BR>
benefit from a 199.<BR>
<BR>
This is one of those cases that could probably benefit from simulation<BR>
to judge the merits. My gut feel is that the costs of the UAS sending<BR>
the 199 outweigh the benefits. Barring some analysis to the contrary I<BR>
think I would say that a UAS SHOULD NOT send a 199. Or perhaps MAY send,<BR>
but without any guidance of when.<BR>
<BR>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Paul<BR>
<BR>
Dale.Worley@comcast.net wrote:<BR>
&gt; Here are some comments on this I-D.&nbsp; I don't think that any of th=
em<BR>
&gt; modify our understanding of the proposal.&nbsp; I guess this is very t=
erse;<BR>
&gt; please use these suggestions as you see fit.<BR>
&gt;<BR>
&gt; * Abstract<BR>
&gt;<BR>
&gt; Clarify:<BR>
&gt;<BR>
&gt;&nbsp;&nbsp;&nbsp; This specification defines a new SIP response code, =
199 Early<BR>
&gt;&nbsp;&nbsp;&nbsp; Dialog Terminated, which a SIP entity can use to ind=
icate upstream<BR>
&gt;&nbsp;&nbsp;&nbsp; that an early dialog has been terminated.&nbsp; The =
response code can be<BR>
&gt;&nbsp;&nbsp;&nbsp; used by a SIP entity (a SIP proxy or UAS), to inform=
 the UAC that<BR>
&gt;&nbsp;&nbsp;&nbsp; an early dialog has been terminated.&nbsp; As the 19=
9 response is a<BR>
&gt;&nbsp;&nbsp;&nbsp; provisional response, often the UAC will receive it =
before<BR>
&gt;&nbsp;&nbsp;&nbsp; receiving a final response to its request.<BR>
&gt;<BR>
&gt; * Section 1<BR>
&gt;<BR>
&gt;&nbsp;&nbsp;&nbsp; When a forking proxy receives a non-2xx final respon=
se, it does not<BR>
&gt;&nbsp;&nbsp;&nbsp; always immediately forward the response upstream tow=
ards the sender<BR>
&gt;&nbsp;&nbsp;&nbsp; of the associated request.&nbsp; Instead, the forkin=
g proxy &quot;stores&quot; it<BR>
&gt;&nbsp;&nbsp;&nbsp; and waits for further final responses from remote de=
stinations where<BR>
&gt;&nbsp;&nbsp;&nbsp; the forked request was forwarded.<BR>
&gt;<BR>
&gt; The text does not allow for serial forking, only parallel forking,<BR>
&gt; which might be confusing to people who are more aware of serial<BR>
&gt; forking.&nbsp; So amend the last sentence to:<BR>
&gt;<BR>
&gt;&nbsp;&nbsp;&nbsp; Instead, the forking proxy &quot;stores&quot; it and=
 waits for further final<BR>
&gt;&nbsp;&nbsp;&nbsp; responses from remote destinations where the forked =
request was<BR>
&gt;&nbsp;&nbsp;&nbsp; forwarded, and possibly forwards the request to addi=
tional<BR>
&gt;&nbsp;&nbsp;&nbsp; destinations.<BR>
&gt;<BR>
&gt; Also, Avoid using &quot;...&quot; except when indicating that the word=
 is used<BR>
&gt; in a non-standard way or that one is discussing the word itself:<BR>
&gt;<BR>
&gt;&nbsp;&nbsp;&nbsp; Instead, the forking proxy &quot;stores&quot; it [..=
.]<BR>
&gt;<BR>
&gt; should be:<BR>
&gt;<BR>
&gt;&nbsp;&nbsp;&nbsp; Instead, the forking proxy stores it [...]<BR>
&gt;<BR>
&gt; * Section 3.<BR>
&gt;<BR>
&gt;&nbsp;&nbsp;&nbsp; REQ 1: It must be possible to indicate to the UAC th=
at an early<BR>
&gt;&nbsp;&nbsp;&nbsp; dialog has been terminated before a final response i=
s sent.<BR>
&gt;<BR>
&gt; Oddly, though everyone thinks they understand this sentence, it<BR>
&gt; doesn't really say what we want.&nbsp; To really ensure that the UAC<B=
R>
&gt; understands the early dialog has been terminated before the UAS sends<=
BR>
&gt; a final response, the UAS would have to send 199 *and get the PRACK<BR>
&gt; saying that it was received* before sending a final response.&nbsp; Th=
is may<BR>
&gt; well be worse than the present state of affairs.<BR>
&gt;<BR>
&gt; What we really mean is:<BR>
&gt;<BR>
&gt;&nbsp;&nbsp;&nbsp; REQ 1: It must be possible for a UAS or proxy to sen=
d an indication<BR>
&gt;&nbsp;&nbsp;&nbsp; to the UAC that an early dialog has been terminated,=
 which<BR>
&gt;&nbsp;&nbsp;&nbsp; indication will not be delayed waiting for a final r=
esponse from<BR>
&gt;&nbsp;&nbsp;&nbsp; any UAS.<BR>
&gt;<BR>
&gt; * Section 4<BR>
&gt;<BR>
&gt;&nbsp;&nbsp;&nbsp; When a client receives a 199 Early Dialog Terminated=
 provisional<BR>
&gt;&nbsp;&nbsp;&nbsp; response it MAY release resources and procedures ass=
ociated with the<BR>
&gt;&nbsp;&nbsp;&nbsp; early dialog the 199 response is received on.<BR>
&gt;<BR>
&gt; Amend to clarify the meaning of 199 independently of the consequences<=
BR>
&gt; of that meaning:<BR>
&gt;<BR>
&gt;&nbsp;&nbsp;&nbsp; When a client receives a 199 Early Dialog Terminated=
 provisional<BR>
&gt;&nbsp;&nbsp;&nbsp; response, it has positive knowledge that the UAS tha=
t created the<BR>
&gt;&nbsp;&nbsp;&nbsp; early dialog has terminated it, and so the client MA=
Y release<BR>
&gt;&nbsp;&nbsp;&nbsp; resources and procedures associated with the early d=
ialog for which<BR>
&gt;&nbsp;&nbsp;&nbsp; the 199 response is received.<BR>
&gt;<BR>
&gt; * Section 4<BR>
&gt;<BR>
&gt; Add this paragraph before the last paragraph:<BR>
&gt;<BR>
&gt;&nbsp;&nbsp;&nbsp; A UAC that receives a 199 response for an early dial=
og MUST NOT<BR>
&gt;&nbsp;&nbsp;&nbsp; send any further requests on that dialog, except for=
 a PRACK if the<BR>
&gt;&nbsp;&nbsp;&nbsp; 199 response was sent reliably, and PRACKs for other=
 reliable<BR>
&gt;&nbsp;&nbsp;&nbsp; provisional responses that are to be acknowledged.<B=
R>
&gt;<BR>
&gt; This is copied from section 7 (Backward Compatibility), but since it<B=
R>
&gt; is a constraint on UACs, it should also be stated in this section.<BR>
&gt;<BR>
&gt; * Section 4<BR>
&gt;<BR>
&gt;&nbsp;&nbsp;&nbsp; The UAC MAY use additional information (for example =
the final<BR>
&gt;&nbsp;&nbsp;&nbsp; response which triggered the 199 response) received =
in the 199<BR>
&gt;&nbsp;&nbsp;&nbsp; response when initiating new sessions, if it is poss=
ible to avoid the<BR>
&gt;&nbsp;&nbsp;&nbsp; new session initiation request to be rejected.<BR>
&gt;<BR>
&gt; What is the meaning of the final clause?<BR>
&gt;<BR>
&gt; * Section 5<BR>
&gt;<BR>
&gt; Add &quot;early&quot; as marked:<BR>
&gt;<BR>
&gt;&nbsp;&nbsp;&nbsp; When a UAS wants to terminate *an early* dialog it s=
ends a non-2xx<BR>
&gt;&nbsp;&nbsp;&nbsp; SIP final response, as specified in [RFC3261].&nbsp;=
 In addition, prior<BR>
&gt;&nbsp;&nbsp;&nbsp; to sending the non-2xx SIP final response, the UAS M=
AY send a 199<BR>
&gt;&nbsp;&nbsp;&nbsp; response to indicate that the dialog is being termin=
ated.<BR>
&gt;<BR>
&gt; This clarifies that 199 cannot be used to terminate a confirmed<BR>
&gt; dialog.<BR>
&gt;<BR>
&gt; (The reason a UAS may want to send a 199 is to provide this<BR>
&gt; information even when the upstream proxy does not support 199.)<BR>
&gt;<BR>
&gt; There is a subtlty to the second sentence:<BR>
&gt;<BR>
&gt; Sending the 199 *before* the final response is not necessary due to<BR>
&gt; the semantics of 199 (as it will probably reach the UAC first anyway,<=
BR>
&gt; and it would not matter if it did not), but rather to prevent the<BR>
&gt; upstream proxy from generating a 199 based on the final response, and<=
BR>
&gt; then to transmit the UAS's 199 immediately after -- if the UAS sends<B=
R>
&gt; the 199 first, any upstream proxy will (most likely) see the 199<BR>
&gt; before seeing the final response (which might induce the proxy to send=
<BR>
&gt; its own 199).<BR>
&gt;<BR>
&gt; * Section 6<BR>
&gt;<BR>
&gt;&nbsp;&nbsp;&nbsp; When a forking proxy receives a non-2xx final respon=
se which<BR>
&gt;&nbsp;&nbsp;&nbsp; terminates one or more early dialogs and the proxy d=
oes not intend to<BR>
&gt;&nbsp;&nbsp;&nbsp; forward the final response immediately (due to the r=
ules for a<BR>
&gt;&nbsp;&nbsp;&nbsp; forking proxy) it MUST send a 199 provisional respon=
se, for each<BR>
&gt;&nbsp;&nbsp;&nbsp; associated early dialog that it can associate with t=
he final<BR>
&gt;&nbsp;&nbsp;&nbsp; response, upstream towards the sender of the associa=
ted request.<BR>
&gt;<BR>
&gt; Clarify, and note that if the proxy has already passed a 199 for an<BR>
&gt; early dialog, it need not generate one itself.&nbsp; Also, I've reduce=
d<BR>
&gt; MUST to SHOULD, because (as the following NOTE explains), the proxy<BR>
&gt; may not know of all early dialogs which are terminated by the final<BR>
&gt; response.<BR>
&gt;<BR>
&gt;&nbsp;&nbsp;&nbsp; [...] it SHOULD initiate a 199 provisional response =
upstream<BR>
&gt;&nbsp;&nbsp;&nbsp; towards the sender of the associated request, for ea=
ch terminated<BR>
&gt;&nbsp;&nbsp;&nbsp; early dialog, excepting those for which it has alrea=
dy transmitted<BR>
&gt;&nbsp;&nbsp;&nbsp; a 199 response.<BR>
&gt;<BR>
&gt; Note that the &quot;already transmitted 199 responses&quot; may have c=
ome from<BR>
&gt; downstream proxies as well as downstream UASs, so this consideration<B=
R>
&gt; applies even if we do not allow UASs to send 199s.<BR>
&gt;<BR>
&gt; * Section 7<BR>
&gt;<BR>
&gt;&nbsp;&nbsp;&nbsp; Since all SIP entities involved in a session setup d=
o not necessarily<BR>
&gt;&nbsp;&nbsp;&nbsp; support the specific meaning of the 199 Early Dialog=
 Terminated<BR>
&gt;&nbsp;&nbsp;&nbsp; provisional response, the sender of the response MUS=
T be prepared to<BR>
&gt;&nbsp;&nbsp;&nbsp; receive SIP requests associated with the dialog for =
which the 199<BR>
&gt;&nbsp;&nbsp;&nbsp; response was sent.<BR>
&gt;<BR>
&gt; This situation can also result due to race conditions.&nbsp; Amend to:=
<BR>
&gt;<BR>
&gt;&nbsp;&nbsp;&nbsp; Because of race conditions and because the client ma=
y not support the<BR>
&gt;&nbsp;&nbsp;&nbsp; specific meaning of the 199 provisional response, th=
e sender of the<BR>
&gt;&nbsp;&nbsp;&nbsp; response MUST be prepared to receive SIP requests as=
sociated with<BR>
&gt;&nbsp;&nbsp;&nbsp; the dialog for which the 199 response was sent.<BR>
&gt;<BR>
&gt; And add a qualifier to this sentence:<BR>
&gt;<BR>
&gt;&nbsp;&nbsp;&nbsp; If such request is received, and the receiver mainta=
ins state of<BR>
&gt;&nbsp;&nbsp;&nbsp; the dialogs, the receiver MUST reply to such request=
s with a 481<BR>
&gt;&nbsp;&nbsp;&nbsp; final response (unless another error response is mor=
e appropriate).<BR>
&gt;<BR>
&gt; * Section 7<BR>
&gt;<BR>
&gt; Remove &quot;...&quot; from this sentence and add &quot;even&quot; as =
marked:<BR>
&gt;<BR>
&gt;&nbsp;&nbsp;&nbsp; The 199 Early Dialog Terminated response code MUST N=
OT replace a<BR>
&gt;&nbsp;&nbsp;&nbsp; final response.&nbsp; A final response MUST always b=
e sent, *even* after<BR>
&gt;&nbsp;&nbsp;&nbsp; one or many 199 Early Dialog Terminated provisional =
responses have<BR>
&gt;&nbsp;&nbsp;&nbsp; been sent.<BR>
&gt;<BR>
&gt; * Section 8<BR>
&gt;<BR>
&gt; Clarify this sentence to:<BR>
&gt;<BR>
&gt;&nbsp;&nbsp;&nbsp; The 199 Early Dialog Terminated response code allows=
 a SIP entity<BR>
&gt;&nbsp;&nbsp;&nbsp; to indicate to upstream entities that a specific ear=
ly dialog has<BR>
&gt;&nbsp;&nbsp;&nbsp; been terminated, before a final response reaches the=
 upstream<BR>
&gt;&nbsp;&nbsp;&nbsp; entities.<BR>
&gt;<BR>
&gt; Add the following:<BR>
&gt;<BR>
&gt;&nbsp;&nbsp;&nbsp; The early dialog that has been terminated is identif=
ied by the Call-Id<BR>
&gt;&nbsp;&nbsp;&nbsp; header and the to-tag value of the 199 response.<BR>
&gt;<BR>
&gt;&nbsp;&nbsp;&nbsp; Future standards may define optional additional info=
rmation carried<BR>
&gt;&nbsp;&nbsp;&nbsp; in the headers and/or body of the response.<BR>
&gt;<BR>
&gt; * Section 9<BR>
&gt;<BR>
&gt;&nbsp;&nbsp;&nbsp; A 199 Early Dialog Terminated provisional response M=
UST NOT contain a<BR>
&gt;&nbsp;&nbsp;&nbsp; new SDP offer/answer message body, but the sender of=
 the response MAY<BR>
&gt;&nbsp;&nbsp;&nbsp; insert a copy of a previously sent offer/answer mess=
age body as<BR>
&gt;&nbsp;&nbsp;&nbsp; otherwise allowed by the offer/answer rules for a pr=
ovisional<BR>
&gt;&nbsp;&nbsp;&nbsp; response.<BR>
&gt;<BR>
&gt; This rule seems excessively strict, as it seems OK to send a new SDP<B=
R>
&gt; answer in a 199, even if the dialog will be immediately terminated.<BR>
&gt; Certainly, the UAC cannot tell the difference, as there could have<BR>
&gt; been a previous but lost 1xx response carrying the same SDP.<BR>
&gt;<BR>
&gt; The interaction of 199 and SDP probably needs further study...<BR>
&gt;<BR>
&gt; * Section 11<BR>
&gt;<BR>
&gt; Nit:&nbsp; Use &quot;reject&quot; here rather than &quot;rejects&quot;=
:<BR>
&gt;<BR>
&gt;&nbsp;&nbsp;&nbsp; UAS_2 and UAS_3 *reject* the INVITE by sending a 4xx=
 error response.<BR>
&gt;<BR>
&gt; Here is a modification of the example, showing how a UAS can send a<BR>
&gt; 199, and how the proxy handles it.&nbsp; (You may or may not want to u=
se<BR>
&gt; this.)&nbsp; I've also added the ACK's for the 4xx's, which were missi=
ng in<BR>
&gt; the original.<BR>
&gt;<BR>
&gt;&nbsp;&nbsp;&nbsp; [...] UAS_2 and UAS_3 reject the INVITE by sending a=
 4xx error<BR>
&gt;&nbsp;&nbsp;&nbsp; response.&nbsp; UAS_3 sends a 199 with its 4xx.&nbsp=
; When P1 receives the<BR>
&gt;&nbsp;&nbsp;&nbsp; 4xx response from UAS_2 it immediately sends a 199 E=
arly Dialog<BR>
&gt;&nbsp;&nbsp;&nbsp; Terminated associated with UAS_2's dialog towards th=
e UAC.&nbsp; P1<BR>
&gt;&nbsp;&nbsp;&nbsp; transmits UAS_3's 199, and so does not initiate one =
itself for<BR>
&gt;&nbsp;&nbsp;&nbsp; UAS_3's dialog when it receives UAS_3's 4xx response=
.<BR>
&gt;<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; UAC&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; P1&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp; UAS_2&nbsp;&nbsp;&nbsp; UAS_3&nbsp;&nbsp;&nbsp; UAS_4<BR>
&gt;<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; --- INVITE ------&gt;<BR>
&gt;&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; --- INVITE (leg 2) -&gt;<BR>
&gt;&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; --- INVITE (leg 3) ----------&gt;<BR>
&gt;&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; --- INVITE (leg 4) -------------------&gt;<BR>
&gt;&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; &lt;-- 18x (leg 2) -----<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;-- 18x (leg 2) --<BR>
&gt;&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; &lt;-- 18x (leg 3) --------------<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;-- 18x (leg 3) --<BR>
&gt;&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; &lt;-- 18x (leg 4) -----------------------<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;-- 18x (leg 4) --<BR>
&gt;&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; &lt;-- 4xx (leg 2) -----<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;-- 199 (leg 2) --<BR>
&gt;&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; --- ACK (leg 2) ----&gt;<BR>
&gt;&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; &lt;-- 199 (leg 3) --------------<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;-- 199 (leg 3) --<BR>
&gt;&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; &lt;-- 4xx (leg 3) --------------<BR>
&gt;&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; --- ACK (leg 3) -------------&gt;<BR>
&gt;&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; &lt;-- 200 (leg 4) -----------------------<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;-- 200 (leg 4) --<BR>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; --- ACK (leg 4) -&gt;<BR>
&gt;&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; --- ACK (leg 4) ----------------------&gt;<BR>
&gt;<BR>
&gt; Dale<BR>
&gt; _______________________________________________<BR>
&gt; Sip mailing list&nbsp; <A HREF=3D"https://www.ietf.org/mailman/listinf=
o/sip">https://www.ietf.org/mailman/listinfo/sip</A><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>
&gt;<BR>
_______________________________________________<BR>
Sip mailing list&nbsp; <A HREF=3D"https://www.ietf.org/mailman/listinfo/sip=
">https://www.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>
</FONT>
</P>

<br><br>
<P align=3Dleft><FONT style=3D"BACKGROUND-COLOR: #ffffff">.</FONT></P>
</body>
</HTML>=

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_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
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Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
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Attila,

Sure we can take the approach of leaving it to the UAS to decide whether 
to send a 199.

My concern is that it may be inefficient, not just for that UA, but for 
the system as a whole. So I think we need to be a little careful here.

I'm not suggesting forbidding this - clearly there are cases where it is 
the right thing to do.

	Paul

Attila Sipos wrote:
> 
>  >>However I am still scratching my head about the UAS sending a 199. While
>  >>I see no reason to forbid it I'm trying to find a situation when it
>  >>would be helpful.
> 
> It may or may not be helpful but as you say, you can
> see no reason to forbid it.
> 
> Why restrict it if the restriction isn't needed?
> 
> As someone has pointed out, a B2BUA scenario might benefit
> from 199.  In fact, any kind of SIP gateway might benefit.
> 
>  > This is one of those cases that could probably benefit from
>  > simulation to judge the merits. My gut feel is that the costs
>  > of the UAS sending the 199 outweigh the benefits.
> 
> If the UAS wants to send it, that is a decision for
> them - not something that needs to be prevented.
> 
> SIP forking, in the technical sense of sending a
> response with a different to-tag, is always legal.
> So, any UA which has used forking
> should also be able to use 199.
> 
> I don't think we need to include any specific text about UAS
> except to say that a UAS can use it if the UAS has created
> the forking.
> 
> Regards,
> 
> Attila
> 
> 
> 
> -----Original Message-----
> From: sip-bounces@ietf.org on behalf of Paul Kyzivat
> Sent: Fri 13/06/2008 15:07
> To: Dale.Worley@comcast.net
> Cc: sip@ietf.org
> Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
> 
> This all sounds like good stuff.
> 
> However I am still scratching my head about the UAS sending a 199. While
> I see no reason to forbid it I'm trying to find a situation when it
> would be helpful. The only case I have been able to think of is:
> 
>    There is a forking proxy upstream of the UAS, but it doesn't
>    support 199. Yet it may delay the final response while it
>    tries other forks. So the UAS sending the 199 makes up for
>    the inadequacy of the forking proxy.
> 
> The trouble with this is that it is inefficient in many cases. If each
> proxy that forks also supports the sending of 199, and the UAS doesn't
> send one, then we avoid sending the message except in cases where it is
> helpful. If the UAS always sends it we have increased the message count
> by one for all failing calls. And the majority of failing calls get no
> benefit from a 199.
> 
> This is one of those cases that could probably benefit from simulation
> to judge the merits. My gut feel is that the costs of the UAS sending
> the 199 outweigh the benefits. Barring some analysis to the contrary I
> think I would say that a UAS SHOULD NOT send a 199. Or perhaps MAY send,
> but without any guidance of when.
> 
>         Paul
> 
> Dale.Worley@comcast.net wrote:
>  > Here are some comments on this I-D.  I don't think that any of them
>  > modify our understanding of the proposal.  I guess this is very terse;
>  > please use these suggestions as you see fit.
>  >
>  > * Abstract
>  >
>  > Clarify:
>  >
>  >    This specification defines a new SIP response code, 199 Early
>  >    Dialog Terminated, which a SIP entity can use to indicate upstream
>  >    that an early dialog has been terminated.  The response code can be
>  >    used by a SIP entity (a SIP proxy or UAS), to inform the UAC that
>  >    an early dialog has been terminated.  As the 199 response is a
>  >    provisional response, often the UAC will receive it before
>  >    receiving a final response to its request.
>  >
>  > * Section 1
>  >
>  >    When a forking proxy receives a non-2xx final response, it does not
>  >    always immediately forward the response upstream towards the sender
>  >    of the associated request.  Instead, the forking proxy "stores" it
>  >    and waits for further final responses from remote destinations where
>  >    the forked request was forwarded.
>  >
>  > The text does not allow for serial forking, only parallel forking,
>  > which might be confusing to people who are more aware of serial
>  > forking.  So amend the last sentence to:
>  >
>  >    Instead, the forking proxy "stores" it and waits for further final
>  >    responses from remote destinations where the forked request was
>  >    forwarded, and possibly forwards the request to additional
>  >    destinations.
>  >
>  > Also, Avoid using "..." except when indicating that the word is used
>  > in a non-standard way or that one is discussing the word itself:
>  >
>  >    Instead, the forking proxy "stores" it [...]
>  >
>  > should be:
>  >
>  >    Instead, the forking proxy stores it [...]
>  >
>  > * Section 3.
>  >
>  >    REQ 1: It must be possible to indicate to the UAC that an early
>  >    dialog has been terminated before a final response is sent.
>  >
>  > Oddly, though everyone thinks they understand this sentence, it
>  > doesn't really say what we want.  To really ensure that the UAC
>  > understands the early dialog has been terminated before the UAS sends
>  > a final response, the UAS would have to send 199 *and get the PRACK
>  > saying that it was received* before sending a final response.  This may
>  > well be worse than the present state of affairs.
>  >
>  > What we really mean is:
>  >
>  >    REQ 1: It must be possible for a UAS or proxy to send an indication
>  >    to the UAC that an early dialog has been terminated, which
>  >    indication will not be delayed waiting for a final response from
>  >    any UAS.
>  >
>  > * Section 4
>  >
>  >    When a client receives a 199 Early Dialog Terminated provisional
>  >    response it MAY release resources and procedures associated with the
>  >    early dialog the 199 response is received on.
>  >
>  > Amend to clarify the meaning of 199 independently of the consequences
>  > of that meaning:
>  >
>  >    When a client receives a 199 Early Dialog Terminated provisional
>  >    response, it has positive knowledge that the UAS that created the
>  >    early dialog has terminated it, and so the client MAY release
>  >    resources and procedures associated with the early dialog for which
>  >    the 199 response is received.
>  >
>  > * Section 4
>  >
>  > Add this paragraph before the last paragraph:
>  >
>  >    A UAC that receives a 199 response for an early dialog MUST NOT
>  >    send any further requests on that dialog, except for a PRACK if the
>  >    199 response was sent reliably, and PRACKs for other reliable
>  >    provisional responses that are to be acknowledged.
>  >
>  > This is copied from section 7 (Backward Compatibility), but since it
>  > is a constraint on UACs, it should also be stated in this section.
>  >
>  > * Section 4
>  >
>  >    The UAC MAY use additional information (for example the final
>  >    response which triggered the 199 response) received in the 199
>  >    response when initiating new sessions, if it is possible to avoid the
>  >    new session initiation request to be rejected.
>  >
>  > What is the meaning of the final clause?
>  >
>  > * Section 5
>  >
>  > Add "early" as marked:
>  >
>  >    When a UAS wants to terminate *an early* dialog it sends a non-2xx
>  >    SIP final response, as specified in [RFC3261].  In addition, prior
>  >    to sending the non-2xx SIP final response, the UAS MAY send a 199
>  >    response to indicate that the dialog is being terminated.
>  >
>  > This clarifies that 199 cannot be used to terminate a confirmed
>  > dialog.
>  >
>  > (The reason a UAS may want to send a 199 is to provide this
>  > information even when the upstream proxy does not support 199.)
>  >
>  > There is a subtlty to the second sentence:
>  >
>  > Sending the 199 *before* the final response is not necessary due to
>  > the semantics of 199 (as it will probably reach the UAC first anyway,
>  > and it would not matter if it did not), but rather to prevent the
>  > upstream proxy from generating a 199 based on the final response, and
>  > then to transmit the UAS's 199 immediately after -- if the UAS sends
>  > the 199 first, any upstream proxy will (most likely) see the 199
>  > before seeing the final response (which might induce the proxy to send
>  > its own 199).
>  >
>  > * Section 6
>  >
>  >    When a forking proxy receives a non-2xx final response which
>  >    terminates one or more early dialogs and the proxy does not intend to
>  >    forward the final response immediately (due to the rules for a
>  >    forking proxy) it MUST send a 199 provisional response, for each
>  >    associated early dialog that it can associate with the final
>  >    response, upstream towards the sender of the associated request.
>  >
>  > Clarify, and note that if the proxy has already passed a 199 for an
>  > early dialog, it need not generate one itself.  Also, I've reduced
>  > MUST to SHOULD, because (as the following NOTE explains), the proxy
>  > may not know of all early dialogs which are terminated by the final
>  > response.
>  >
>  >    [...] it SHOULD initiate a 199 provisional response upstream
>  >    towards the sender of the associated request, for each terminated
>  >    early dialog, excepting those for which it has already transmitted
>  >    a 199 response.
>  >
>  > Note that the "already transmitted 199 responses" may have come from
>  > downstream proxies as well as downstream UASs, so this consideration
>  > applies even if we do not allow UASs to send 199s.
>  >
>  > * Section 7
>  >
>  >    Since all SIP entities involved in a session setup do not necessarily
>  >    support the specific meaning of the 199 Early Dialog Terminated
>  >    provisional response, the sender of the response MUST be prepared to
>  >    receive SIP requests associated with the dialog for which the 199
>  >    response was sent.
>  >
>  > This situation can also result due to race conditions.  Amend to:
>  >
>  >    Because of race conditions and because the client may not support the
>  >    specific meaning of the 199 provisional response, the sender of the
>  >    response MUST be prepared to receive SIP requests associated with
>  >    the dialog for which the 199 response was sent.
>  >
>  > And add a qualifier to this sentence:
>  >
>  >    If such request is received, and the receiver maintains state of
>  >    the dialogs, the receiver MUST reply to such requests with a 481
>  >    final response (unless another error response is more appropriate).
>  >
>  > * Section 7
>  >
>  > Remove "..." from this sentence and add "even" as marked:
>  >
>  >    The 199 Early Dialog Terminated response code MUST NOT replace a
>  >    final response.  A final response MUST always be sent, *even* after
>  >    one or many 199 Early Dialog Terminated provisional responses have
>  >    been sent.
>  >
>  > * Section 8
>  >
>  > Clarify this sentence to:
>  >
>  >    The 199 Early Dialog Terminated response code allows a SIP entity
>  >    to indicate to upstream entities that a specific early dialog has
>  >    been terminated, before a final response reaches the upstream
>  >    entities.
>  >
>  > Add the following:
>  >
>  >    The early dialog that has been terminated is identified by the Call-Id
>  >    header and the to-tag value of the 199 response.
>  >
>  >    Future standards may define optional additional information carried
>  >    in the headers and/or body of the response.
>  >
>  > * Section 9
>  >
>  >    A 199 Early Dialog Terminated provisional response MUST NOT contain a
>  >    new SDP offer/answer message body, but the sender of the response MAY
>  >    insert a copy of a previously sent offer/answer message body as
>  >    otherwise allowed by the offer/answer rules for a provisional
>  >    response.
>  >
>  > This rule seems excessively strict, as it seems OK to send a new SDP
>  > answer in a 199, even if the dialog will be immediately terminated.
>  > Certainly, the UAC cannot tell the difference, as there could have
>  > been a previous but lost 1xx response carrying the same SDP.
>  >
>  > The interaction of 199 and SDP probably needs further study...
>  >
>  > * Section 11
>  >
>  > Nit:  Use "reject" here rather than "rejects":
>  >
>  >    UAS_2 and UAS_3 *reject* the INVITE by sending a 4xx error response.
>  >
>  > Here is a modification of the example, showing how a UAS can send a
>  > 199, and how the proxy handles it.  (You may or may not want to use
>  > this.)  I've also added the ACK's for the 4xx's, which were missing in
>  > the original.
>  >
>  >    [...] UAS_2 and UAS_3 reject the INVITE by sending a 4xx error
>  >    response.  UAS_3 sends a 199 with its 4xx.  When P1 receives the
>  >    4xx response from UAS_2 it immediately sends a 199 Early Dialog
>  >    Terminated associated with UAS_2's dialog towards the UAC.  P1
>  >    transmits UAS_3's 199, and so does not initiate one itself for
>  >    UAS_3's dialog when it receives UAS_3's 4xx response.
>  >
>  >       UAC               P1                  UAS_2    UAS_3    UAS_4
>  >
>  >        --- INVITE ------>
>  >                          --- INVITE (leg 2) ->
>  >                          --- INVITE (leg 3) ---------->
>  >                          --- INVITE (leg 4) ------------------->
>  >                          <-- 18x (leg 2) -----
>  >        <-- 18x (leg 2) --
>  >                          <-- 18x (leg 3) --------------
>  >        <-- 18x (leg 3) --
>  >                          <-- 18x (leg 4) -----------------------
>  >        <-- 18x (leg 4) --
>  >                          <-- 4xx (leg 2) -----
>  >        <-- 199 (leg 2) --
>  >                          --- ACK (leg 2) ---->
>  >                          <-- 199 (leg 3) --------------
>  >        <-- 199 (leg 3) --
>  >                          <-- 4xx (leg 3) --------------
>  >                          --- ACK (leg 3) ------------->
>  >                          <-- 200 (leg 4) -----------------------
>  >        <-- 200 (leg 4) --
>  >        --- ACK (leg 4) ->
>  >                          --- ACK (leg 4) ---------------------->
>  >
>  > Dale
>  > _______________________________________________
>  > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>  > This list is for NEW development of the core SIP Protocol
>  > Use sip-implementors@cs.columbia.edu for questions on current sip
>  > Use sipping@ietf.org for new developments on the application of sip
>  >
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
> 
> 
> .
> 
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>,
	"Attila Sipos" <attila.sipos@vegastream.com>
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Hi,
 
Maybe we could talk about the case when a UAS establishes more than one dialog? 
 
Regards,
 
Christer

________________________________

From: sip-bounces@ietf.org on behalf of Paul Kyzivat
Sent: Sat 14/06/2008 23:35
To: Attila Sipos
Cc: sip@ietf.org
Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00



Attila,

Sure we can take the approach of leaving it to the UAS to decide whether
to send a 199.

My concern is that it may be inefficient, not just for that UA, but for
the system as a whole. So I think we need to be a little careful here.

I'm not suggesting forbidding this - clearly there are cases where it is
the right thing to do.

        Paul

Attila Sipos wrote:
>
>  >>However I am still scratching my head about the UAS sending a 199. While
>  >>I see no reason to forbid it I'm trying to find a situation when it
>  >>would be helpful.
>
> It may or may not be helpful but as you say, you can
> see no reason to forbid it.
>
> Why restrict it if the restriction isn't needed?
>
> As someone has pointed out, a B2BUA scenario might benefit
> from 199.  In fact, any kind of SIP gateway might benefit.
>
>  > This is one of those cases that could probably benefit from
>  > simulation to judge the merits. My gut feel is that the costs
>  > of the UAS sending the 199 outweigh the benefits.
>
> If the UAS wants to send it, that is a decision for
> them - not something that needs to be prevented.
>
> SIP forking, in the technical sense of sending a
> response with a different to-tag, is always legal.
> So, any UA which has used forking
> should also be able to use 199.
>
> I don't think we need to include any specific text about UAS
> except to say that a UAS can use it if the UAS has created
> the forking.
>
> Regards,
>
> Attila
>
>
>
> -----Original Message-----
> From: sip-bounces@ietf.org on behalf of Paul Kyzivat
> Sent: Fri 13/06/2008 15:07
> To: Dale.Worley@comcast.net
> Cc: sip@ietf.org
> Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
>
> This all sounds like good stuff.
>
> However I am still scratching my head about the UAS sending a 199. While
> I see no reason to forbid it I'm trying to find a situation when it
> would be helpful. The only case I have been able to think of is:
>
>    There is a forking proxy upstream of the UAS, but it doesn't
>    support 199. Yet it may delay the final response while it
>    tries other forks. So the UAS sending the 199 makes up for
>    the inadequacy of the forking proxy.
>
> The trouble with this is that it is inefficient in many cases. If each
> proxy that forks also supports the sending of 199, and the UAS doesn't
> send one, then we avoid sending the message except in cases where it is
> helpful. If the UAS always sends it we have increased the message count
> by one for all failing calls. And the majority of failing calls get no
> benefit from a 199.
>
> This is one of those cases that could probably benefit from simulation
> to judge the merits. My gut feel is that the costs of the UAS sending
> the 199 outweigh the benefits. Barring some analysis to the contrary I
> think I would say that a UAS SHOULD NOT send a 199. Or perhaps MAY send,
> but without any guidance of when.
>
>         Paul
>
> Dale.Worley@comcast.net wrote:
>  > Here are some comments on this I-D.  I don't think that any of them
>  > modify our understanding of the proposal.  I guess this is very terse;
>  > please use these suggestions as you see fit.
>  >
>  > * Abstract
>  >
>  > Clarify:
>  >
>  >    This specification defines a new SIP response code, 199 Early
>  >    Dialog Terminated, which a SIP entity can use to indicate upstream
>  >    that an early dialog has been terminated.  The response code can be
>  >    used by a SIP entity (a SIP proxy or UAS), to inform the UAC that
>  >    an early dialog has been terminated.  As the 199 response is a
>  >    provisional response, often the UAC will receive it before
>  >    receiving a final response to its request.
>  >
>  > * Section 1
>  >
>  >    When a forking proxy receives a non-2xx final response, it does not
>  >    always immediately forward the response upstream towards the sender
>  >    of the associated request.  Instead, the forking proxy "stores" it
>  >    and waits for further final responses from remote destinations where
>  >    the forked request was forwarded.
>  >
>  > The text does not allow for serial forking, only parallel forking,
>  > which might be confusing to people who are more aware of serial
>  > forking.  So amend the last sentence to:
>  >
>  >    Instead, the forking proxy "stores" it and waits for further final
>  >    responses from remote destinations where the forked request was
>  >    forwarded, and possibly forwards the request to additional
>  >    destinations.
>  >
>  > Also, Avoid using "..." except when indicating that the word is used
>  > in a non-standard way or that one is discussing the word itself:
>  >
>  >    Instead, the forking proxy "stores" it [...]
>  >
>  > should be:
>  >
>  >    Instead, the forking proxy stores it [...]
>  >
>  > * Section 3.
>  >
>  >    REQ 1: It must be possible to indicate to the UAC that an early
>  >    dialog has been terminated before a final response is sent.
>  >
>  > Oddly, though everyone thinks they understand this sentence, it
>  > doesn't really say what we want.  To really ensure that the UAC
>  > understands the early dialog has been terminated before the UAS sends
>  > a final response, the UAS would have to send 199 *and get the PRACK
>  > saying that it was received* before sending a final response.  This may
>  > well be worse than the present state of affairs.
>  >
>  > What we really mean is:
>  >
>  >    REQ 1: It must be possible for a UAS or proxy to send an indication
>  >    to the UAC that an early dialog has been terminated, which
>  >    indication will not be delayed waiting for a final response from
>  >    any UAS.
>  >
>  > * Section 4
>  >
>  >    When a client receives a 199 Early Dialog Terminated provisional
>  >    response it MAY release resources and procedures associated with the
>  >    early dialog the 199 response is received on.
>  >
>  > Amend to clarify the meaning of 199 independently of the consequences
>  > of that meaning:
>  >
>  >    When a client receives a 199 Early Dialog Terminated provisional
>  >    response, it has positive knowledge that the UAS that created the
>  >    early dialog has terminated it, and so the client MAY release
>  >    resources and procedures associated with the early dialog for which
>  >    the 199 response is received.
>  >
>  > * Section 4
>  >
>  > Add this paragraph before the last paragraph:
>  >
>  >    A UAC that receives a 199 response for an early dialog MUST NOT
>  >    send any further requests on that dialog, except for a PRACK if the
>  >    199 response was sent reliably, and PRACKs for other reliable
>  >    provisional responses that are to be acknowledged.
>  >
>  > This is copied from section 7 (Backward Compatibility), but since it
>  > is a constraint on UACs, it should also be stated in this section.
>  >
>  > * Section 4
>  >
>  >    The UAC MAY use additional information (for example the final
>  >    response which triggered the 199 response) received in the 199
>  >    response when initiating new sessions, if it is possible to avoid the
>  >    new session initiation request to be rejected.
>  >
>  > What is the meaning of the final clause?
>  >
>  > * Section 5
>  >
>  > Add "early" as marked:
>  >
>  >    When a UAS wants to terminate *an early* dialog it sends a non-2xx
>  >    SIP final response, as specified in [RFC3261].  In addition, prior
>  >    to sending the non-2xx SIP final response, the UAS MAY send a 199
>  >    response to indicate that the dialog is being terminated.
>  >
>  > This clarifies that 199 cannot be used to terminate a confirmed
>  > dialog.
>  >
>  > (The reason a UAS may want to send a 199 is to provide this
>  > information even when the upstream proxy does not support 199.)
>  >
>  > There is a subtlty to the second sentence:
>  >
>  > Sending the 199 *before* the final response is not necessary due to
>  > the semantics of 199 (as it will probably reach the UAC first anyway,
>  > and it would not matter if it did not), but rather to prevent the
>  > upstream proxy from generating a 199 based on the final response, and
>  > then to transmit the UAS's 199 immediately after -- if the UAS sends
>  > the 199 first, any upstream proxy will (most likely) see the 199
>  > before seeing the final response (which might induce the proxy to send
>  > its own 199).
>  >
>  > * Section 6
>  >
>  >    When a forking proxy receives a non-2xx final response which
>  >    terminates one or more early dialogs and the proxy does not intend to
>  >    forward the final response immediately (due to the rules for a
>  >    forking proxy) it MUST send a 199 provisional response, for each
>  >    associated early dialog that it can associate with the final
>  >    response, upstream towards the sender of the associated request.
>  >
>  > Clarify, and note that if the proxy has already passed a 199 for an
>  > early dialog, it need not generate one itself.  Also, I've reduced
>  > MUST to SHOULD, because (as the following NOTE explains), the proxy
>  > may not know of all early dialogs which are terminated by the final
>  > response.
>  >
>  >    [...] it SHOULD initiate a 199 provisional response upstream
>  >    towards the sender of the associated request, for each terminated
>  >    early dialog, excepting those for which it has already transmitted
>  >    a 199 response.
>  >
>  > Note that the "already transmitted 199 responses" may have come from
>  > downstream proxies as well as downstream UASs, so this consideration
>  > applies even if we do not allow UASs to send 199s.
>  >
>  > * Section 7
>  >
>  >    Since all SIP entities involved in a session setup do not necessarily
>  >    support the specific meaning of the 199 Early Dialog Terminated
>  >    provisional response, the sender of the response MUST be prepared to
>  >    receive SIP requests associated with the dialog for which the 199
>  >    response was sent.
>  >
>  > This situation can also result due to race conditions.  Amend to:
>  >
>  >    Because of race conditions and because the client may not support the
>  >    specific meaning of the 199 provisional response, the sender of the
>  >    response MUST be prepared to receive SIP requests associated with
>  >    the dialog for which the 199 response was sent.
>  >
>  > And add a qualifier to this sentence:
>  >
>  >    If such request is received, and the receiver maintains state of
>  >    the dialogs, the receiver MUST reply to such requests with a 481
>  >    final response (unless another error response is more appropriate).
>  >
>  > * Section 7
>  >
>  > Remove "..." from this sentence and add "even" as marked:
>  >
>  >    The 199 Early Dialog Terminated response code MUST NOT replace a
>  >    final response.  A final response MUST always be sent, *even* after
>  >    one or many 199 Early Dialog Terminated provisional responses have
>  >    been sent.
>  >
>  > * Section 8
>  >
>  > Clarify this sentence to:
>  >
>  >    The 199 Early Dialog Terminated response code allows a SIP entity
>  >    to indicate to upstream entities that a specific early dialog has
>  >    been terminated, before a final response reaches the upstream
>  >    entities.
>  >
>  > Add the following:
>  >
>  >    The early dialog that has been terminated is identified by the Call-Id
>  >    header and the to-tag value of the 199 response.
>  >
>  >    Future standards may define optional additional information carried
>  >    in the headers and/or body of the response.
>  >
>  > * Section 9
>  >
>  >    A 199 Early Dialog Terminated provisional response MUST NOT contain a
>  >    new SDP offer/answer message body, but the sender of the response MAY
>  >    insert a copy of a previously sent offer/answer message body as
>  >    otherwise allowed by the offer/answer rules for a provisional
>  >    response.
>  >
>  > This rule seems excessively strict, as it seems OK to send a new SDP
>  > answer in a 199, even if the dialog will be immediately terminated.
>  > Certainly, the UAC cannot tell the difference, as there could have
>  > been a previous but lost 1xx response carrying the same SDP.
>  >
>  > The interaction of 199 and SDP probably needs further study...
>  >
>  > * Section 11
>  >
>  > Nit:  Use "reject" here rather than "rejects":
>  >
>  >    UAS_2 and UAS_3 *reject* the INVITE by sending a 4xx error response.
>  >
>  > Here is a modification of the example, showing how a UAS can send a
>  > 199, and how the proxy handles it.  (You may or may not want to use
>  > this.)  I've also added the ACK's for the 4xx's, which were missing in
>  > the original.
>  >
>  >    [...] UAS_2 and UAS_3 reject the INVITE by sending a 4xx error
>  >    response.  UAS_3 sends a 199 with its 4xx.  When P1 receives the
>  >    4xx response from UAS_2 it immediately sends a 199 Early Dialog
>  >    Terminated associated with UAS_2's dialog towards the UAC.  P1
>  >    transmits UAS_3's 199, and so does not initiate one itself for
>  >    UAS_3's dialog when it receives UAS_3's 4xx response.
>  >
>  >       UAC               P1                  UAS_2    UAS_3    UAS_4
>  >
>  >        --- INVITE ------>
>  >                          --- INVITE (leg 2) ->
>  >                          --- INVITE (leg 3) ---------->
>  >                          --- INVITE (leg 4) ------------------->
>  >                          <-- 18x (leg 2) -----
>  >        <-- 18x (leg 2) --
>  >                          <-- 18x (leg 3) --------------
>  >        <-- 18x (leg 3) --
>  >                          <-- 18x (leg 4) -----------------------
>  >        <-- 18x (leg 4) --
>  >                          <-- 4xx (leg 2) -----
>  >        <-- 199 (leg 2) --
>  >                          --- ACK (leg 2) ---->
>  >                          <-- 199 (leg 3) --------------
>  >        <-- 199 (leg 3) --
>  >                          <-- 4xx (leg 3) --------------
>  >                          --- ACK (leg 3) ------------->
>  >                          <-- 200 (leg 4) -----------------------
>  >        <-- 200 (leg 4) --
>  >        --- ACK (leg 4) ->
>  >                          --- ACK (leg 4) ---------------------->
>  >
>  > Dale
>  > _______________________________________________
>  > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>  > This list is for NEW development of the core SIP Protocol
>  > Use sip-implementors@cs.columbia.edu for questions on current sip
>  > Use sipping@ietf.org for new developments on the application of sip
>  >
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>
>
>
> .
>
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


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


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From: Rockson Zhengyu Li <lzyiii@laszlomail.com>
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Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt
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Hi,

one questions on this draft.

Since RFC3261 says, non-100 1xx response for INVITE cause early-dialog=20
creation.
does this draft will normativly change the criteria, excluding 199?

If, the answer is Yes, I think it need be stated explicitly in the=20
draft.

thanks

Regards,
-Rockson

On Sat, May 24, 2008 at  1:15 AM, 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=20
Group of the IETF.

=09Title=09=09: Response Code for Indication of Terminated Dialog
=09Author(s)=09: C. Holmberg
=09Filename=09: draft-ietf-sip-199-00.txt
=09Pages=09=09: 9
=09Date=09=09: 2008-5-15
=09
This specification defines a new SIP response code, 199 Early Dialog
=C2=A0=C2=A0 Terminated, which a SIP entity can use to indicate upstream th=
at an
=C2=A0=C2=A0 early dialog has been terminated.=C2=A0 The response code can =
be used by a
=C2=A0=C2=A0 SIP entity, normally a forking SIP proxy, to allow the UAC to =
know
=C2=A0=C2=A0 that an early dialog has been terminated before it a final res=
ponse
=C2=A0=C2=A0 is sent to the request.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-199-00.txt=20
<http://www.ietf.org/internet-drafts/draft-ietf-sip-199-00.txt> =C2=A0=20
<http://www.ietf.org/internet-drafts/draft-ietf-sip-199-00.txt>

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/=20
<ftp://ftp.ietf.org/internet-drafts/> =C2=A0=20
<ftp://ftp.ietf.org/internet-drafts/>

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>
> ------------------------------
>
> _______________________________________________
Sip mailing list=C2=A0 https://www.ietf.org/mailman/listinfo/sip=20
<https://www.ietf.org/mailman/listinfo/sip> =C2=A0=20
<https://www.ietf.org/mailman/listinfo/sip>
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu=20
<mailto:sip-implementors@cs.columbia.edu> =C2=A0=20
<mailto:sip-implementors@cs.columbia.edu> =C2=A0for questions on current si=
p
Use sipping@ietf.org <mailto:sipping@ietf.org> =C2=A0=20
<mailto:sipping@ietf.org> =C2=A0for new developments on the application of=
=20
sip

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<DIV ALIGN=3D"LEFT"><FONT FACE=3D"Verdana" SIZE=3D"2" COLOR=3D"#000000" LET=
TERSPACING=3D"0" KERNING=3D"0">Hi,</FONT></DIV><FONT FACE=3D"Verdana" SIZE=
=3D"2" COLOR=3D"#000000" LETTERSPACING=3D"0" KERNING=3D"0"><br></FONT><DIV =
ALIGN=3D"LEFT"><FONT FACE=3D"Verdana" SIZE=3D"2" COLOR=3D"#000000" LETTERSP=
ACING=3D"0" KERNING=3D"0">one questions on this draft.</FONT></DIV><FONT FA=
CE=3D"Verdana" SIZE=3D"2" COLOR=3D"#000000" LETTERSPACING=3D"0" KERNING=3D"=
0"><br></FONT><DIV ALIGN=3D"LEFT"><FONT FACE=3D"Verdana" SIZE=3D"2" COLOR=
=3D"#000000" LETTERSPACING=3D"0" KERNING=3D"0">Since RFC3261 says, non-100 =
1xx response for INVITE cause early-dialog creation.</FONT></DIV><DIV ALIGN=
=3D"LEFT"><FONT FACE=3D"Verdana" SIZE=3D"2" COLOR=3D"#000000" LETTERSPACING=
=3D"0" KERNING=3D"0">does this draft will normativly change the criteria, e=
xcluding 199?</FONT></DIV><FONT FACE=3D"Verdana" SIZE=3D"2" COLOR=3D"#00000=
0" LETTERSPACING=3D"0" KERNING=3D"0"><br></FONT><DIV ALIGN=3D"LEFT"><FONT F=
ACE=3D"Verdana" SIZE=3D"2" COLOR=3D"#000000" LETTERSPACING=3D"0" KERNING=3D=
"0">If, the answer is Yes, I think it need be stated explicitly in the draf=
t.</FONT></DIV><FONT FACE=3D"Verdana" SIZE=3D"2" COLOR=3D"#000000" LETTERSP=
ACING=3D"0" KERNING=3D"0"><br></FONT><FONT FACE=3D"Verdana" SIZE=3D"2" COLO=
R=3D"#000000" LETTERSPACING=3D"0" KERNING=3D"0"><br></FONT><DIV ALIGN=3D"LE=
FT"><FONT FACE=3D"Verdana" SIZE=3D"2" COLOR=3D"#000000" LETTERSPACING=3D"0"=
 KERNING=3D"0">thanks</FONT></DIV><FONT FACE=3D"Verdana" SIZE=3D"2" COLOR=
=3D"#000000" LETTERSPACING=3D"0" KERNING=3D"0"><br></FONT><DIV ALIGN=3D"LEF=
T"><FONT FACE=3D"Verdana" SIZE=3D"2" COLOR=3D"#000000" LETTERSPACING=3D"0" =
KERNING=3D"0">Regards,</FONT></DIV><DIV ALIGN=3D"LEFT"><FONT FACE=3D"Verdan=
a" SIZE=3D"2" COLOR=3D"#000000" LETTERSPACING=3D"0" KERNING=3D"0">-Rockson&=
nbsp; </FONT></DIV><FONT FACE=3D"Verdana" SIZE=3D"2" COLOR=3D"#000000" LETT=
ERSPACING=3D"0" KERNING=3D"0"><br></FONT><FONT FACE=3D"Verdana" SIZE=3D"2" =
COLOR=3D"#000000" LETTERSPACING=3D"0" KERNING=3D"0"><br></FONT><DIV ALIGN=
=3D"LEFT"><FONT FACE=3D"Verdana" SIZE=3D"2" COLOR=3D"#000000" LETTERSPACING=
=3D"0" KERNING=3D"0">On Sat, May 24, 2008 at&nbsp; 1:15 AM, Internet-Drafts=
@ietf.org wrote:</FONT></DIV><FONT FACE=3D"Verdana" SIZE=3D"2" COLOR=3D"#00=
0000" LETTERSPACING=3D"0" KERNING=3D"0"><br></FONT><DIV ALIGN=3D"LEFT"><FON=
T FACE=3D"Verdana" SIZE=3D"2" COLOR=3D"#000000" LETTERSPACING=3D"0" KERNING=
=3D"0">&gt; A New Internet-Draft is available from the on-line Internet-Dra=
fts </FONT></DIV><DIV ALIGN=3D"LEFT"><FONT FACE=3D"Verdana" SIZE=3D"2" COLO=
R=3D"#000000" LETTERSPACING=3D"0" KERNING=3D"0">directories.</FONT></DIV><D=
IV ALIGN=3D"LEFT"><FONT FACE=3D"Verdana" SIZE=3D"2" COLOR=3D"#000000" LETTE=
RSPACING=3D"0" KERNING=3D"0">This draft is a work item of the Session Initi=
ation Protocol Working Group of the IETF.</FONT></DIV><FONT FACE=3D"Verdana=
" SIZE=3D"2" COLOR=3D"#000000" LETTERSPACING=3D"0" KERNING=3D"0"><br></FONT=
><DIV ALIGN=3D"LEFT"><FONT FACE=3D"Verdana" SIZE=3D"2" COLOR=3D"#000000" LE=
TTERSPACING=3D"0" KERNING=3D"0">=09Title=09=09: Response Code for Indicatio=
n of Terminated Dialog</FONT></DIV><DIV ALIGN=3D"LEFT"><FONT FACE=3D"Verdan=
a" SIZE=3D"2" COLOR=3D"#000000" LETTERSPACING=3D"0" KERNING=3D"0">=09Author=
(s)=09: C. Holmberg</FONT></DIV><DIV ALIGN=3D"LEFT"><FONT FACE=3D"Verdana" =
SIZE=3D"2" COLOR=3D"#000000" LETTERSPACING=3D"0" KERNING=3D"0">=09Filename=
=09: draft-ietf-sip-199-00.txt</FONT></DIV><DIV ALIGN=3D"LEFT"><FONT FACE=
=3D"Verdana" SIZE=3D"2" COLOR=3D"#000000" LETTERSPACING=3D"0" KERNING=3D"0"=
>=09Pages=09=09: 9</FONT></DIV><DIV ALIGN=3D"LEFT"><FONT FACE=3D"Verdana" S=
IZE=3D"2" COLOR=3D"#000000" LETTERSPACING=3D"0" KERNING=3D"0">=09Date=09=09=
: 2008-5-15</FONT></DIV><DIV ALIGN=3D"LEFT"><FONT FACE=3D"Verdana" SIZE=3D"=
2" COLOR=3D"#000000" LETTERSPACING=3D"0" KERNING=3D"0">=09</FONT></DIV><DIV=
 ALIGN=3D"LEFT"><FONT FACE=3D"Verdana" SIZE=3D"2" COLOR=3D"#000000" LETTERS=
PACING=3D"0" KERNING=3D"0">This specification defines a new SIP response co=
de, 199 Early Dialog</FONT></DIV><DIV ALIGN=3D"LEFT"><FONT FACE=3D"Verdana"=
 SIZE=3D"2" COLOR=3D"#000000" LETTERSPACING=3D"0" KERNING=3D"0">=C2=A0=C2=
=A0 Terminated, which a SIP entity can use to indicate upstream that an</FO=
NT></DIV><DIV ALIGN=3D"LEFT"><FONT FACE=3D"Verdana" SIZE=3D"2" COLOR=3D"#00=
0000" LETTERSPACING=3D"0" KERNING=3D"0">=C2=A0=C2=A0 early dialog has been =
terminated.=C2=A0 The response code can be used by a</FONT></DIV><DIV ALIGN=
=3D"LEFT"><FONT FACE=3D"Verdana" SIZE=3D"2" COLOR=3D"#000000" LETTERSPACING=
=3D"0" KERNING=3D"0">=C2=A0=C2=A0 SIP entity, normally a forking SIP proxy,=
 to allow the UAC to know</FONT></DIV><DIV ALIGN=3D"LEFT"><FONT FACE=3D"Ver=
dana" SIZE=3D"2" COLOR=3D"#000000" LETTERSPACING=3D"0" KERNING=3D"0">=C2=A0=
=C2=A0 that an early dialog has been terminated before it a final response<=
/FONT></DIV><DIV ALIGN=3D"LEFT"><FONT FACE=3D"Verdana" SIZE=3D"2" COLOR=3D"=
#000000" LETTERSPACING=3D"0" KERNING=3D"0">=C2=A0=C2=A0 is sent to the requ=
est.</FONT></DIV><FONT FACE=3D"Verdana" SIZE=3D"2" COLOR=3D"#000000" LETTER=
SPACING=3D"0" KERNING=3D"0"><br></FONT><DIV ALIGN=3D"LEFT"><FONT FACE=3D"Ve=
rdana" SIZE=3D"2" COLOR=3D"#000000" LETTERSPACING=3D"0" KERNING=3D"0">A URL=
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HREF=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-199-00.txt" TARG=
ET=3D"_blank"><U>http://www.ietf.org/internet-drafts/draft-ietf-sip-199-00.=
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II version of the</FONT></DIV><DIV ALIGN=3D"LEFT"><FONT FACE=3D"Verdana" SI=
ZE=3D"2" COLOR=3D"#000000" LETTERSPACING=3D"0" KERNING=3D"0">Internet-Draft=
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na" SIZE=3D"2" COLOR=3D"#000000" LETTERSPACING=3D"0" KERNING=3D"0">&gt; </F=
ONT></DIV><DIV ALIGN=3D"LEFT"><FONT FACE=3D"Verdana" SIZE=3D"2" COLOR=3D"#0=
00000" LETTERSPACING=3D"0" KERNING=3D"0">&gt; -----------------------------=
-</FONT></DIV><DIV ALIGN=3D"LEFT"><FONT FACE=3D"Verdana" SIZE=3D"2" COLOR=
=3D"#000000" LETTERSPACING=3D"0" KERNING=3D"0">&gt; </FONT></DIV><DIV ALIGN=
=3D"LEFT"><FONT FACE=3D"Verdana" SIZE=3D"2" COLOR=3D"#000000" LETTERSPACING=
=3D"0" KERNING=3D"0">&gt; _______________________________________________</=
FONT></DIV><DIV ALIGN=3D"LEFT"><FONT FACE=3D"Verdana" SIZE=3D"2" COLOR=3D"#=
000000" LETTERSPACING=3D"0" KERNING=3D"0">Sip mailing list=C2=A0 <FONT COLO=
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FACE=3D"Verdana" SIZE=3D"2" COLOR=3D"#000000" LETTERSPACING=3D"0" KERNING=
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V><DIV ALIGN=3D"LEFT"><FONT FACE=3D"Verdana" SIZE=3D"2" COLOR=3D"#000000" L=
ETTERSPACING=3D"0" KERNING=3D"0">Use <FONT COLOR=3D"#3333FF"><A HREF=3D"mai=
lto:sip-implementors@cs.columbia.edu"><U>sip-implementors@cs.columbia.edu</=
U></A><FONT COLOR=3D"#000000"><A HREF=3D"mailto:sip-implementors@cs.columbi=
a.edu">=C2=A0</A>=C2=A0for questions on current sip</FONT></FONT></FONT></D=
IV><DIV ALIGN=3D"LEFT"><FONT FACE=3D"Verdana" SIZE=3D"2" COLOR=3D"#000000" =
LETTERSPACING=3D"0" KERNING=3D"0">Use <FONT COLOR=3D"#3333FF"><A HREF=3D"ma=
ilto:sipping@ietf.org"><U>sipping@ietf.org</U></A><FONT COLOR=3D"#000000"><=
A HREF=3D"mailto:sipping@ietf.org">=C2=A0</A>=C2=A0for new developments on =
the application of sip</FONT></FONT></FONT></DIV>
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_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============0405690635==--


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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Rockson Zhengyu Li" <lzyiii@laszlomail.com>,
	<sip@ietf.org>
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Hi,
=20
You wouldn't send a 199 unless you have already sent a non-100 1xx =
response, so the early dialog has already been created.
=20
Regards,
=20
Christer

________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =
Rockson Zhengyu Li
Sent: 15. kes=E4kuuta 2008 14:43
To: sip@ietf.org
Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt


Hi,


one questions on this draft.


Since RFC3261 says, non-100 1xx response for INVITE cause early-dialog =
creation.
does this draft will normativly change the criteria, excluding 199?


If, the answer is Yes, I think it need be stated explicitly in the =
draft.



thanks


Regards,
-Rockson =20



On Sat, May 24, 2008 at  1:15 AM, Internet-Drafts@ietf.org wrote:


> A New Internet-Draft is available from the on-line Internet-Drafts=20
directories.
This draft is a work item of the Session Initiation Protocol Working =
Group of the IETF.


Title : Response Code for Indication of Terminated Dialog
Author(s) : C. Holmberg
Filename : draft-ietf-sip-199-00.txt
Pages : 9
Date : 2008-5-15
This specification defines a new SIP response code, 199 Early Dialog
   Terminated, which a SIP entity can use to indicate upstream that an
   early dialog has been terminated.  The response code can be used by a
   SIP entity, normally a forking SIP proxy, to allow the UAC to know
   that an early dialog has been terminated before it a final response
   is sent to the request.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-199-00.txt =
<http://www.ietf.org/internet-drafts/draft-ietf-sip-199-00.txt>   =
<http://www.ietf.org/internet-drafts/draft-ietf-sip-199-00.txt>=20


Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/ =
<ftp://ftp.ietf.org/internet-drafts/>   =
<ftp://ftp.ietf.org/internet-drafts/>=20


Below is the data which will enable a MIME compliant mail reader
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Internet-Draft.


>=20
> ------------------------------
>=20
> _______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip =
<https://www.ietf.org/mailman/listinfo/sip>   =
<https://www.ietf.org/mailman/listinfo/sip>=20
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu =
<mailto:sip-implementors@cs.columbia.edu>   =
<mailto:sip-implementors@cs.columbia.edu>  for questions on current sip
Use sipping@ietf.org <mailto:sipping@ietf.org>   =
<mailto:sipping@ietf.org>  for new developments on the application of =
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
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<BODY>
<DIV><SPAN class=3D500285111-15062008><FONT face=3DArial color=3D#0000ff =

size=3D2>Hi,</FONT></SPAN></DIV>
<DIV><SPAN class=3D500285111-15062008><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D500285111-15062008><FONT face=3DArial color=3D#0000ff =
size=3D2>You=20
wouldn't send a 199 unless you have already sent a non-100 1xx response, =
so the=20
early dialog has already been created.</FONT></SPAN></DIV>
<DIV><SPAN class=3D500285111-15062008><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D500285111-15062008><FONT face=3DArial color=3D#0000ff =

size=3D2>Regards,</FONT></SPAN></DIV>
<DIV><SPAN class=3D500285111-15062008><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D500285111-15062008><FONT face=3DArial color=3D#0000ff =

size=3D2>Christer</FONT></SPAN></DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
[mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Rockson Zhengyu=20
Li<BR><B>Sent:</B> 15. kes=E4kuuta 2008 14:43<BR><B>To:</B>=20
sip@ietf.org<BR><B>Subject:</B> Re: [Sip] I-D=20
ACTION:draft-ietf-sip-199-00.txt<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">Hi,</FONT></DIV><FONT face=3DVerdana color=3D#000000 =
size=3D2=20
KERNING=3D"0" LETTERSPACING=3D"0"><BR></FONT>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">one questions on this draft.</FONT></DIV><FONT =
face=3DVerdana=20
color=3D#000000 size=3D2 KERNING=3D"0" LETTERSPACING=3D"0"><BR></FONT>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">Since RFC3261 says, non-100 1xx response for INVITE =
cause=20
early-dialog creation.</FONT></DIV>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">does this draft will normativly change the criteria, =
excluding=20
199?</FONT></DIV><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0"><BR></FONT>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">If, the answer is Yes, I think it need be stated =
explicitly in=20
the draft.</FONT></DIV><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0"><BR></FONT><FONT face=3DVerdana color=3D#000000 =
size=3D2 KERNING=3D"0"=20
LETTERSPACING=3D"0"><BR></FONT>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">thanks</FONT></DIV><FONT face=3DVerdana =
color=3D#000000 size=3D2=20
KERNING=3D"0" LETTERSPACING=3D"0"><BR></FONT>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">Regards,</FONT></DIV>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">-Rockson&nbsp; </FONT></DIV><FONT face=3DVerdana =
color=3D#000000=20
size=3D2 KERNING=3D"0" LETTERSPACING=3D"0"><BR></FONT><FONT =
face=3DVerdana color=3D#000000=20
size=3D2 KERNING=3D"0" LETTERSPACING=3D"0"><BR></FONT>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">On Sat, May 24, 2008 at&nbsp; 1:15 AM,=20
Internet-Drafts@ietf.org wrote:</FONT></DIV><FONT face=3DVerdana =
color=3D#000000=20
size=3D2 KERNING=3D"0" LETTERSPACING=3D"0"><BR></FONT>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">&gt; A New Internet-Draft is available from the =
on-line=20
Internet-Drafts </FONT></DIV>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">directories.</FONT></DIV>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">This draft is a work item of the Session Initiation =
Protocol=20
Working Group of the IETF.</FONT></DIV><FONT face=3DVerdana =
color=3D#000000 size=3D2=20
KERNING=3D"0" LETTERSPACING=3D"0"><BR></FONT>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">Title : Response Code for Indication of Terminated=20
Dialog</FONT></DIV>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">Author(s) : C. Holmberg</FONT></DIV>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">Filename : draft-ietf-sip-199-00.txt</FONT></DIV>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">Pages : 9</FONT></DIV>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">Date : 2008-5-15</FONT></DIV>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0"></FONT></DIV>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">This specification defines a new SIP response code, =
199 Early=20
Dialog</FONT></DIV>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">&nbsp;&nbsp; Terminated, which a SIP entity can use =
to=20
indicate upstream that an</FONT></DIV>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">&nbsp;&nbsp; early dialog has been terminated.&nbsp; =
The=20
response code can be used by a</FONT></DIV>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">&nbsp;&nbsp; SIP entity, normally a forking SIP =
proxy, to=20
allow the UAC to know</FONT></DIV>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">&nbsp;&nbsp; that an early dialog has been =
terminated before=20
it a final response</FONT></DIV>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">&nbsp;&nbsp; is sent to the =
request.</FONT></DIV><FONT=20
face=3DVerdana color=3D#000000 size=3D2 KERNING=3D"0" =
LETTERSPACING=3D"0"><BR></FONT>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">A URL for this Internet-Draft is:</FONT></DIV>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#3333ff size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0"><A=20
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-199-00.txt"=20
target=3D_blank><U>http://www.ietf.org/internet-drafts/draft-ietf-sip-199=
-00.txt</U></A><FONT=20
color=3D#000000><A=20
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-sip-199-00.txt"=20
target=3D_blank>&nbsp;</A></FONT></FONT></DIV><FONT face=3DVerdana =
color=3D#000000=20
size=3D2 KERNING=3D"0" LETTERSPACING=3D"0"><BR></FONT>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">Internet-Drafts are also available by anonymous FTP=20
at:</FONT></DIV>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#3333ff size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0"><A href=3D"ftp://ftp.ietf.org/internet-drafts/"=20
target=3D_blank><U>ftp://ftp.ietf.org/internet-drafts/</U></A><FONT=20
color=3D#000000><A href=3D"ftp://ftp.ietf.org/internet-drafts/"=20
target=3D_blank>&nbsp;</A></FONT></FONT></DIV><FONT face=3DVerdana =
color=3D#000000=20
size=3D2 KERNING=3D"0" LETTERSPACING=3D"0"><BR></FONT>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">Below is the data which will enable a MIME compliant =
mail=20
reader</FONT></DIV>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">implementation to automatically retrieve the ASCII =
version of=20
the</FONT></DIV>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">Internet-Draft.</FONT></DIV><FONT face=3DVerdana =
color=3D#000000=20
size=3D2 KERNING=3D"0" LETTERSPACING=3D"0"><BR></FONT>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">&gt; </FONT></DIV>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">&gt; ------------------------------</FONT></DIV>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">&gt; </FONT></DIV>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">&gt;=20
_______________________________________________</FONT></DIV>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">Sip mailing list&nbsp; <FONT color=3D#3333ff><A=20
href=3D"https://www.ietf.org/mailman/listinfo/sip"=20
target=3D_blank><U>https://www.ietf.org/mailman/listinfo/sip</U></A><FONT=
=20
color=3D#000000><A href=3D"https://www.ietf.org/mailman/listinfo/sip"=20
target=3D_blank>&nbsp;</A></FONT></FONT></FONT></DIV>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">This list is for NEW development of the core SIP=20
Protocol</FONT></DIV>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#000000 size=3D2 =
KERNING=3D"0"=20
LETTERSPACING=3D"0">Use <FONT color=3D#3333ff><A=20
href=3D"mailto:sip-implementors@cs.columbia.edu"><U>sip-implementors@cs.c=
olumbia.edu</U></A><FONT=20
color=3D#000000><A=20
href=3D"mailto:sip-implementors@cs.columbia.edu">&nbsp;</A>&nbsp;for =
questions on=20
current sip</FONT></FONT></FONT></DIV>
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Hi Christer,
=20
It would be easier for UAS to tell if it sends non-100 1xx response
creating early-dialog before sending 199.
However, it's a little bit difficult from UAC's perspective, since the
non-100 1xx response might not be reliable.
=20
If UAC does not get , say 180, but get 199 first, it looks like it does
not make any sense.
=20
One option is to mandate usage of 100rel, which might be too strict for
this draft.
=20
Regards,
-Rockson
------------------------------------------------------------------------
-------------------
Date: Sun, 15 Jun 2008 13:52:24 +0200
From: "Christer Holmberg" <christer.holmberg@ericsson.com>
Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt
To: "Rockson Zhengyu Li" <lzyiii@laszlomail.com>, <sip@ietf.org>
Message-ID:
 <CA9998CD4A020D418654FCDEF4E707DF06B19A2E@esealmw113.eemea.ericsson.se>
=20
Content-Type: text/plain; charset=3D"iso-8859-1"
=20
Hi,
=20
You wouldn't send a 199 unless you have already sent a non-100 1xx
response, so the early dialog has already been created.
=20
Regards,
=20
Christer
=20
________________________________
=20
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Rockson Zhengyu Li
Sent: 15. kes?kuuta 2008 14:43
To: sip@ietf.org
Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt
=20

Hi,
=20

one questions on this draft.
=20

Since RFC3261 says, non-100 1xx response for INVITE cause early-dialog
creation.
does this draft will normativly change the criteria, excluding 199?
=20

If, the answer is Yes, I think it need be stated explicitly in the
draft.
=20
=20
=20
thanks
=20

Regards,
-Rockson =20
=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 6.00.6000.16640" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D390394401-16062008>Hi=20
</SPAN></FONT><FONT face=3DArial size=3D2><SPAN=20
class=3D390394401-16062008>Christer,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D390394401-16062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D390394401-16062008>It =
would be easier=20
for UAS to tell if it sends non-100 1xx response creating early-dialog =
before=20
sending 199.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN =
class=3D390394401-16062008>However,&nbsp;it's=20
a&nbsp;little bit difficult from</SPAN></FONT><FONT face=3DArial =
size=3D2><SPAN=20
class=3D390394401-16062008>&nbsp;UAC's perspective, since the non-100 =
1xx response=20
might not be reliable.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D390394401-16062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D390394401-16062008>If UAC =
does not get=20
, say 180, but get 199 first, it looks like it does not make any=20
sense.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D390394401-16062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D390394401-16062008>One =
option is to=20
mandate usage of 100rel, which might be too strict for this=20
draft.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D390394401-16062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D390394401-16062008>Regards,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D390394401-16062008>-Rockson</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D390394401-16062008>----------------------------------------------=
---------------------------------------------</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Date: Sun, 15 Jun 2008 13:52:24 =
+0200<BR>From:=20
"Christer Holmberg" &lt;<A=20
href=3D"mailto:christer.holmberg@ericsson.com">christer.holmberg@ericsson=
.com</A>&gt;<BR>Subject:=20
Re: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt<BR>To: "Rockson Zhengyu =
Li"=20
&lt;<A=20
href=3D"mailto:lzyiii@laszlomail.com">lzyiii@laszlomail.com</A>&gt;,&nbsp=
;&lt;<A=20
href=3D"mailto:sip@ietf.org">sip@ietf.org</A>&gt;<BR>Message-ID:<BR>&nbsp=
;&lt;<A=20
href=3D"mailto:CA9998CD4A020D418654FCDEF4E707DF06B19A2E@esealmw113.eemea.=
ericsson.se">CA9998CD4A020D418654FCDEF4E707DF06B19A2E@esealmw113.eemea.er=
icsson.se</A>&gt;<BR>&nbsp;<BR>Content-Type:=20
text/plain; charset=3D"iso-8859-1"</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Hi,<BR>&nbsp;<BR>You wouldn't send a =
199 unless you=20
have already sent a non-100 1xx response, so the early dialog has =
already been=20
created.<BR>&nbsp;<BR>Regards,<BR>&nbsp;<BR>Christer</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial =
size=3D2>________________________________</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>From: <A=20
href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</A>=20
[mailto:sip-bounces@ietf.org] On Behalf Of Rockson Zhengyu Li<BR>Sent: =
15.=20
kes?kuuta 2008 14:43<BR>To: <A=20
href=3D"mailto:sip@ietf.org">sip@ietf.org</A><BR>Subject: Re: [Sip] I-D=20
ACTION:draft-ietf-sip-199-00.txt</FONT></DIV>
<DIV>&nbsp;</DIV><FONT face=3DArial size=3D2>
<DIV><BR>Hi,</DIV>
<DIV>&nbsp;</DIV>
<DIV><BR>one questions on this draft.</DIV>
<DIV>&nbsp;</DIV>
<DIV><BR>Since RFC3261 says, non-100 1xx response for INVITE cause =
early-dialog=20
creation.<BR>does this draft will normativly change the criteria, =
excluding=20
199?</DIV>
<DIV>&nbsp;</DIV>
<DIV><BR>If, the answer is Yes, I think it need be stated explicitly in =
the=20
draft.</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>thanks</DIV>
<DIV>&nbsp;</DIV>
<DIV><BR>Regards,<BR>-Rockson&nbsp; </DIV>
<DIV>&nbsp;</DIV>
<DIV></FONT>&nbsp;</DIV></BODY></HTML>

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_______________________________________________
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Use sipping@ietf.org for new developments on the application of sip
--===============1979160237==--


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Referring to introduction part in the draft.
   " At some point the proxy uses a specified mechanism to determine the
"best" final response code, and forwards that final response upstream
towards the sender of the associated request."
=20
Considering a scenario where all the forks have failed in yielding a 2xx
response from different UASes (UAS2 , 3 & 4 in the diagram) with
different error codes, how will the proxy determine the "best" final
response and to which leg would it send the best final response. What
would happen after 199 to leg 4?
=20
      UAC               P1                  UAS_2    UAS_3    UAS_4

       --- INVITE ------>
                         --- INVITE (leg 2) ->
                         --- INVITE (leg 3) ---------->
                         --- INVITE (leg 4) ------------------->
                         <-- 18x (leg 2) -----
       <-- 18x (leg 2) --
                         <-- 18x (leg 3) --------------
       <-- 18x (leg 3) --
                         <-- 18x (leg 4) -----------------------
       <-- 18x (leg 4) --
                         <-- 4xx (leg 2) -----
       <-- 199 (leg 2) --
                         <-- 5xx (leg 3) --------------
       <-- 199 (leg 3) --
                         <-- 6xx (leg 4) -----------------------
       <-- 199 (leg 4) --


Best Regards,=20
Ashish Saxena=20
877-5570=20

=20

________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Attila Sipos
Sent: Saturday, June 14, 2008 9:30 PM
To: Paul Kyzivat; Dale.Worley@comcast.net
Cc: sip@ietf.org
Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00




>>However I am still scratching my head about the UAS sending a 199.
While
>>I see no reason to forbid it I'm trying to find a situation when it
>>would be helpful.

It may or may not be helpful but as you say, you can
see no reason to forbid it.

Why restrict it if the restriction isn't needed?

As someone has pointed out, a B2BUA scenario might benefit
from 199.  In fact, any kind of SIP gateway might benefit.

> This is one of those cases that could probably benefit from
> simulation to judge the merits. My gut feel is that the costs
> of the UAS sending the 199 outweigh the benefits.

If the UAS wants to send it, that is a decision for
them - not something that needs to be prevented.

SIP forking, in the technical sense of sending a
response with a different to-tag, is always legal.
So, any UA which has used forking
should also be able to use 199.

I don't think we need to include any specific text about UAS
except to say that a UAS can use it if the UAS has created
the forking.

Regards,

Attila



-----Original Message-----
From: sip-bounces@ietf.org on behalf of Paul Kyzivat
Sent: Fri 13/06/2008 15:07
To: Dale.Worley@comcast.net
Cc: sip@ietf.org
Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00

This all sounds like good stuff.

However I am still scratching my head about the UAS sending a 199. While
I see no reason to forbid it I'm trying to find a situation when it
would be helpful. The only case I have been able to think of is:

   There is a forking proxy upstream of the UAS, but it doesn't
   support 199. Yet it may delay the final response while it
   tries other forks. So the UAS sending the 199 makes up for
   the inadequacy of the forking proxy.

The trouble with this is that it is inefficient in many cases. If each
proxy that forks also supports the sending of 199, and the UAS doesn't
send one, then we avoid sending the message except in cases where it is
helpful. If the UAS always sends it we have increased the message count
by one for all failing calls. And the majority of failing calls get no
benefit from a 199.

This is one of those cases that could probably benefit from simulation
to judge the merits. My gut feel is that the costs of the UAS sending
the 199 outweigh the benefits. Barring some analysis to the contrary I
think I would say that a UAS SHOULD NOT send a 199. Or perhaps MAY send,
but without any guidance of when.

        Paul

Dale.Worley@comcast.net wrote:
> Here are some comments on this I-D.  I don't think that any of them
> modify our understanding of the proposal.  I guess this is very terse;
> please use these suggestions as you see fit.
>
> * Abstract
>
> Clarify:
>
>    This specification defines a new SIP response code, 199 Early
>    Dialog Terminated, which a SIP entity can use to indicate upstream
>    that an early dialog has been terminated.  The response code can be
>    used by a SIP entity (a SIP proxy or UAS), to inform the UAC that
>    an early dialog has been terminated.  As the 199 response is a
>    provisional response, often the UAC will receive it before
>    receiving a final response to its request.
>
> * Section 1
>
>    When a forking proxy receives a non-2xx final response, it does not
>    always immediately forward the response upstream towards the sender
>    of the associated request.  Instead, the forking proxy "stores" it
>    and waits for further final responses from remote destinations
where
>    the forked request was forwarded.
>
> The text does not allow for serial forking, only parallel forking,
> which might be confusing to people who are more aware of serial
> forking.  So amend the last sentence to:
>
>    Instead, the forking proxy "stores" it and waits for further final
>    responses from remote destinations where the forked request was
>    forwarded, and possibly forwards the request to additional
>    destinations.
>
> Also, Avoid using "..." except when indicating that the word is used
> in a non-standard way or that one is discussing the word itself:
>
>    Instead, the forking proxy "stores" it [...]
>
> should be:
>
>    Instead, the forking proxy stores it [...]
>
> * Section 3.
>
>    REQ 1: It must be possible to indicate to the UAC that an early
>    dialog has been terminated before a final response is sent.
>
> Oddly, though everyone thinks they understand this sentence, it
> doesn't really say what we want.  To really ensure that the UAC
> understands the early dialog has been terminated before the UAS sends
> a final response, the UAS would have to send 199 *and get the PRACK
> saying that it was received* before sending a final response.  This
may
> well be worse than the present state of affairs.
>
> What we really mean is:
>
>    REQ 1: It must be possible for a UAS or proxy to send an indication
>    to the UAC that an early dialog has been terminated, which
>    indication will not be delayed waiting for a final response from
>    any UAS.
>
> * Section 4
>
>    When a client receives a 199 Early Dialog Terminated provisional
>    response it MAY release resources and procedures associated with
the
>    early dialog the 199 response is received on.
>
> Amend to clarify the meaning of 199 independently of the consequences
> of that meaning:
>
>    When a client receives a 199 Early Dialog Terminated provisional
>    response, it has positive knowledge that the UAS that created the
>    early dialog has terminated it, and so the client MAY release
>    resources and procedures associated with the early dialog for which
>    the 199 response is received.
>
> * Section 4
>
> Add this paragraph before the last paragraph:
>
>    A UAC that receives a 199 response for an early dialog MUST NOT
>    send any further requests on that dialog, except for a PRACK if the
>    199 response was sent reliably, and PRACKs for other reliable
>    provisional responses that are to be acknowledged.
>
> This is copied from section 7 (Backward Compatibility), but since it
> is a constraint on UACs, it should also be stated in this section.
>
> * Section 4
>
>    The UAC MAY use additional information (for example the final
>    response which triggered the 199 response) received in the 199
>    response when initiating new sessions, if it is possible to avoid
the
>    new session initiation request to be rejected.
>
> What is the meaning of the final clause?
>
> * Section 5
>
> Add "early" as marked:
>
>    When a UAS wants to terminate *an early* dialog it sends a non-2xx
>    SIP final response, as specified in [RFC3261].  In addition, prior
>    to sending the non-2xx SIP final response, the UAS MAY send a 199
>    response to indicate that the dialog is being terminated.
>
> This clarifies that 199 cannot be used to terminate a confirmed
> dialog.
>
> (The reason a UAS may want to send a 199 is to provide this
> information even when the upstream proxy does not support 199.)
>
> There is a subtlty to the second sentence:
>
> Sending the 199 *before* the final response is not necessary due to
> the semantics of 199 (as it will probably reach the UAC first anyway,
> and it would not matter if it did not), but rather to prevent the
> upstream proxy from generating a 199 based on the final response, and
> then to transmit the UAS's 199 immediately after -- if the UAS sends
> the 199 first, any upstream proxy will (most likely) see the 199
> before seeing the final response (which might induce the proxy to send
> its own 199).
>
> * Section 6
>
>    When a forking proxy receives a non-2xx final response which
>    terminates one or more early dialogs and the proxy does not intend
to
>    forward the final response immediately (due to the rules for a
>    forking proxy) it MUST send a 199 provisional response, for each
>    associated early dialog that it can associate with the final
>    response, upstream towards the sender of the associated request.
>
> Clarify, and note that if the proxy has already passed a 199 for an
> early dialog, it need not generate one itself.  Also, I've reduced
> MUST to SHOULD, because (as the following NOTE explains), the proxy
> may not know of all early dialogs which are terminated by the final
> response.
>
>    [...] it SHOULD initiate a 199 provisional response upstream
>    towards the sender of the associated request, for each terminated
>    early dialog, excepting those for which it has already transmitted
>    a 199 response.
>
> Note that the "already transmitted 199 responses" may have come from
> downstream proxies as well as downstream UASs, so this consideration
> applies even if we do not allow UASs to send 199s.
>
> * Section 7
>
>    Since all SIP entities involved in a session setup do not
necessarily
>    support the specific meaning of the 199 Early Dialog Terminated
>    provisional response, the sender of the response MUST be prepared
to
>    receive SIP requests associated with the dialog for which the 199
>    response was sent.
>
> This situation can also result due to race conditions.  Amend to:
>
>    Because of race conditions and because the client may not support
the
>    specific meaning of the 199 provisional response, the sender of the
>    response MUST be prepared to receive SIP requests associated with
>    the dialog for which the 199 response was sent.
>
> And add a qualifier to this sentence:
>
>    If such request is received, and the receiver maintains state of
>    the dialogs, the receiver MUST reply to such requests with a 481
>    final response (unless another error response is more appropriate).
>
> * Section 7
>
> Remove "..." from this sentence and add "even" as marked:
>
>    The 199 Early Dialog Terminated response code MUST NOT replace a
>    final response.  A final response MUST always be sent, *even* after
>    one or many 199 Early Dialog Terminated provisional responses have
>    been sent.
>
> * Section 8
>
> Clarify this sentence to:
>
>    The 199 Early Dialog Terminated response code allows a SIP entity
>    to indicate to upstream entities that a specific early dialog has
>    been terminated, before a final response reaches the upstream
>    entities.
>
> Add the following:
>
>    The early dialog that has been terminated is identified by the
Call-Id
>    header and the to-tag value of the 199 response.
>
>    Future standards may define optional additional information carried
>    in the headers and/or body of the response.
>
> * Section 9
>
>    A 199 Early Dialog Terminated provisional response MUST NOT contain
a
>    new SDP offer/answer message body, but the sender of the response
MAY
>    insert a copy of a previously sent offer/answer message body as
>    otherwise allowed by the offer/answer rules for a provisional
>    response.
>
> This rule seems excessively strict, as it seems OK to send a new SDP
> answer in a 199, even if the dialog will be immediately terminated.
> Certainly, the UAC cannot tell the difference, as there could have
> been a previous but lost 1xx response carrying the same SDP.
>
> The interaction of 199 and SDP probably needs further study...
>
> * Section 11
>
> Nit:  Use "reject" here rather than "rejects":
>
>    UAS_2 and UAS_3 *reject* the INVITE by sending a 4xx error
response.
>
> Here is a modification of the example, showing how a UAS can send a
> 199, and how the proxy handles it.  (You may or may not want to use
> this.)  I've also added the ACK's for the 4xx's, which were missing in
> the original.
>
>    [...] UAS_2 and UAS_3 reject the INVITE by sending a 4xx error
>    response.  UAS_3 sends a 199 with its 4xx.  When P1 receives the
>    4xx response from UAS_2 it immediately sends a 199 Early Dialog
>    Terminated associated with UAS_2's dialog towards the UAC.  P1
>    transmits UAS_3's 199, and so does not initiate one itself for
>    UAS_3's dialog when it receives UAS_3's 4xx response.
>
>       UAC               P1                  UAS_2    UAS_3    UAS_4
>
>        --- INVITE ------>
>                          --- INVITE (leg 2) ->
>                          --- INVITE (leg 3) ---------->
>                          --- INVITE (leg 4) ------------------->
>                          <-- 18x (leg 2) -----
>        <-- 18x (leg 2) --
>                          <-- 18x (leg 3) --------------
>        <-- 18x (leg 3) --
>                          <-- 18x (leg 4) -----------------------
>        <-- 18x (leg 4) --
>                          <-- 4xx (leg 2) -----
>        <-- 199 (leg 2) --
>                          --- ACK (leg 2) ---->
>                          <-- 199 (leg 3) --------------
>        <-- 199 (leg 3) --
>                          <-- 4xx (leg 3) --------------
>                          --- ACK (leg 3) ------------->
>                          <-- 200 (leg 4) -----------------------
>        <-- 200 (leg 4) --
>        --- ACK (leg 4) ->
>                          --- ACK (leg 4) ---------------------->
>
> Dale
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip





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<HTML><HEAD><TITLE>RE: [Sip] Draft submission: =
draft-ietf-sip-199-00</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.3354" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D937291707-16062008><FONT face=3DVerdana =
color=3D#800000=20
size=3D2>Referring to introduction part in the =
draft.</FONT></SPAN></DIV>
<DIV><SPAN class=3D937291707-16062008>&nbsp;&nbsp;&nbsp;" At some point =
the proxy=20
uses a&nbsp;specified mechanism to determine the "best" final response =
code,=20
and&nbsp;forwards that final response upstream towards the sender of=20
the&nbsp;associated request."</SPAN></DIV>
<DIV><SPAN class=3D937291707-16062008><FONT face=3DVerdana =
color=3D#800000=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D937291707-16062008><FONT face=3DVerdana =
color=3D#800000=20
size=3D2>Considering a scenario where all the forks have failed in =
yielding a 2xx=20
response from different UASes (UAS2 , 3 &amp; 4 in the diagram) with =
different=20
error codes, how will the proxy determine the "best" final response and =
to which=20
leg would it send the best final response. What would happen after 199 =
to leg=20
4?</FONT></SPAN></DIV>
<DIV><SPAN class=3D937291707-16062008><FONT face=3DVerdana =
color=3D#800000=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D937291707-16062008>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
UAC&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
P1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
UAS_2&nbsp;&nbsp;&nbsp; UAS_3&nbsp;&nbsp;&nbsp;=20
UAS_4<BR><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; --- INVITE=20
------&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;=20
--- INVITE (leg 2)=20
-&gt;<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;=20
--- INVITE (leg 3)=20
----------&gt;<BR>&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;=20
--- INVITE (leg 4)=20
-------------------&gt;<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;=20
&lt;-- 18x (leg 2) -----<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;-- =
18x (leg=20
2)=20
--<BR>&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;=20
&lt;-- 18x (leg 3) =
--------------<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;--=20
18x (leg 3)=20
--<BR>&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;=20
&lt;-- 18x (leg 4)=20
-----------------------<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;-- =
18x (leg=20
4)=20
--<BR>&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;=20
&lt;-- <STRONG>4xx</STRONG> (leg 2)=20
-----<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;-- 199 (leg 2)=20
--<BR>&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;=20
&lt;-- <STRONG>5xx </STRONG>(leg 3)=20
--------------<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;-- 199 (leg =
3)=20
--<BR>&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;=20
&lt;--&nbsp;<STRONG>6xx </STRONG>(leg 4)=20
-----------------------<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&lt;--&nbsp;199=20
(leg 4) --<BR></SPAN></DIV><!-- Converted from text/rtf format -->
<P><SPAN lang=3Den-us><FONT face=3DVerdana color=3D#800000 size=3D2>Best =

Regards,</FONT></SPAN> <BR><SPAN lang=3Den-us><FONT face=3DVerdana =
color=3D#800000=20
size=3D2>Ashish Saxena</FONT></SPAN> <BR><SPAN lang=3Den-us><FONT =
face=3DVerdana=20
color=3D#800000 size=3D2>877-5570</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> sip-bounces@ietf.org=20
[mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Attila =
Sipos<BR><B>Sent:</B>=20
Saturday, June 14, 2008 9:30 PM<BR><B>To:</B> Paul Kyzivat;=20
Dale.Worley@comcast.net<BR><B>Cc:</B> sip@ietf.org<BR><B>Subject:</B> =
Re: [Sip]=20
Draft submission: draft-ietf-sip-199-00<BR></FONT><BR></DIV>
<DIV></DIV><!-- Converted from text/plain format --><BR>
<P><FONT size=3D2>&gt;&gt;However I am still scratching my head about =
the UAS=20
sending a 199. While<BR>&gt;&gt;I see no reason to forbid it I'm trying =
to find=20
a situation when it<BR>&gt;&gt;would be helpful.<BR><BR>It may or may =
not be=20
helpful but as you say, you can<BR>see no reason to forbid =
it.<BR><BR>Why=20
restrict it if the restriction isn't needed?<BR><BR>As someone has =
pointed out,=20
a B2BUA scenario might benefit<BR>from 199.&nbsp; In fact, any kind of =
SIP=20
gateway might benefit.<BR><BR>&gt; This is one of those cases that could =

probably benefit from<BR>&gt; simulation to judge the merits. My gut =
feel is=20
that the costs<BR>&gt; of the UAS sending the 199 outweigh the=20
benefits.<BR><BR>If the UAS wants to send it, that is a decision =
for<BR>them -=20
not something that needs to be prevented.<BR><BR>SIP forking, in the =
technical=20
sense of sending a<BR>response with a different to-tag, is always =
legal.<BR>So,=20
any UA which has used forking<BR>should also be able to use =
199.<BR><BR>I don't=20
think we need to include any specific text about UAS<BR>except to say =
that a UAS=20
can use it if the UAS has created<BR>the=20
forking.<BR><BR>Regards,<BR><BR>Attila<BR><BR><BR><BR>-----Original=20
Message-----<BR>From: sip-bounces@ietf.org on behalf of Paul =
Kyzivat<BR>Sent:=20
Fri 13/06/2008 15:07<BR>To: Dale.Worley@comcast.net<BR>Cc:=20
sip@ietf.org<BR>Subject: Re: [Sip] Draft submission:=20
draft-ietf-sip-199-00<BR><BR>This all sounds like good =
stuff.<BR><BR>However I=20
am still scratching my head about the UAS sending a 199. While<BR>I see =
no=20
reason to forbid it I'm trying to find a situation when it<BR>would be =
helpful.=20
The only case I have been able to think of is:<BR><BR>&nbsp;&nbsp; There =
is a=20
forking proxy upstream of the UAS, but it doesn't<BR>&nbsp;&nbsp; =
support 199.=20
Yet it may delay the final response while it<BR>&nbsp;&nbsp; tries other =
forks.=20
So the UAS sending the 199 makes up for<BR>&nbsp;&nbsp; the inadequacy =
of the=20
forking proxy.<BR><BR>The trouble with this is that it is inefficient in =
many=20
cases. If each<BR>proxy that forks also supports the sending of 199, and =
the UAS=20
doesn't<BR>send one, then we avoid sending the message except in cases =
where it=20
is<BR>helpful. If the UAS always sends it we have increased the message=20
count<BR>by one for all failing calls. And the majority of failing calls =
get=20
no<BR>benefit from a 199.<BR><BR>This is one of those cases that could =
probably=20
benefit from simulation<BR>to judge the merits. My gut feel is that the =
costs of=20
the UAS sending<BR>the 199 outweigh the benefits. Barring some analysis =
to the=20
contrary I<BR>think I would say that a UAS SHOULD NOT send a 199. Or =
perhaps MAY=20
send,<BR>but without any guidance of=20
when.<BR><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
Paul<BR><BR>Dale.Worley@comcast.net wrote:<BR>&gt; Here are some =
comments on=20
this I-D.&nbsp; I don't think that any of them<BR>&gt; modify our =
understanding=20
of the proposal.&nbsp; I guess this is very terse;<BR>&gt; please use =
these=20
suggestions as you see fit.<BR>&gt;<BR>&gt; * Abstract<BR>&gt;<BR>&gt;=20
Clarify:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; This specification defines a =
new SIP=20
response code, 199 Early<BR>&gt;&nbsp;&nbsp;&nbsp; Dialog Terminated, =
which a=20
SIP entity can use to indicate upstream<BR>&gt;&nbsp;&nbsp;&nbsp; that =
an early=20
dialog has been terminated.&nbsp; The response code can=20
be<BR>&gt;&nbsp;&nbsp;&nbsp; used by a SIP entity (a SIP proxy or UAS), =
to=20
inform the UAC that<BR>&gt;&nbsp;&nbsp;&nbsp; an early dialog has been=20
terminated.&nbsp; As the 199 response is a<BR>&gt;&nbsp;&nbsp;&nbsp; =
provisional=20
response, often the UAC will receive it before<BR>&gt;&nbsp;&nbsp;&nbsp; =

receiving a final response to its request.<BR>&gt;<BR>&gt; * Section=20
1<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; When a forking proxy receives a =
non-2xx=20
final response, it does not<BR>&gt;&nbsp;&nbsp;&nbsp; always immediately =
forward=20
the response upstream towards the sender<BR>&gt;&nbsp;&nbsp;&nbsp; of =
the=20
associated request.&nbsp; Instead, the forking proxy "stores"=20
it<BR>&gt;&nbsp;&nbsp;&nbsp; and waits for further final responses from =
remote=20
destinations where<BR>&gt;&nbsp;&nbsp;&nbsp; the forked request was=20
forwarded.<BR>&gt;<BR>&gt; The text does not allow for serial forking, =
only=20
parallel forking,<BR>&gt; which might be confusing to people who are =
more aware=20
of serial<BR>&gt; forking.&nbsp; So amend the last sentence=20
to:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; Instead, the forking proxy =
"stores" it and=20
waits for further final<BR>&gt;&nbsp;&nbsp;&nbsp; responses from remote=20
destinations where the forked request was<BR>&gt;&nbsp;&nbsp;&nbsp; =
forwarded,=20
and possibly forwards the request to =
additional<BR>&gt;&nbsp;&nbsp;&nbsp;=20
destinations.<BR>&gt;<BR>&gt; Also, Avoid using "..." except when =
indicating=20
that the word is used<BR>&gt; in a non-standard way or that one is =
discussing=20
the word itself:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; Instead, the forking =
proxy=20
"stores" it [...]<BR>&gt;<BR>&gt; should =
be:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp;=20
Instead, the forking proxy stores it [...]<BR>&gt;<BR>&gt; * Section=20
3.<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; REQ 1: It must be possible to =
indicate to=20
the UAC that an early<BR>&gt;&nbsp;&nbsp;&nbsp; dialog has been =
terminated=20
before a final response is sent.<BR>&gt;<BR>&gt; Oddly, though everyone =
thinks=20
they understand this sentence, it<BR>&gt; doesn't really say what we =
want.&nbsp;=20
To really ensure that the UAC<BR>&gt; understands the early dialog has =
been=20
terminated before the UAS sends<BR>&gt; a final response, the UAS would =
have to=20
send 199 *and get the PRACK<BR>&gt; saying that it was received* before =
sending=20
a final response.&nbsp; This may<BR>&gt; well be worse than the present =
state of=20
affairs.<BR>&gt;<BR>&gt; What we really mean=20
is:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; REQ 1: It must be possible for a =
UAS or=20
proxy to send an indication<BR>&gt;&nbsp;&nbsp;&nbsp; to the UAC that an =
early=20
dialog has been terminated, which<BR>&gt;&nbsp;&nbsp;&nbsp; indication =
will not=20
be delayed waiting for a final response from<BR>&gt;&nbsp;&nbsp;&nbsp; =
any=20
UAS.<BR>&gt;<BR>&gt; * Section 4<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; When =
a client=20
receives a 199 Early Dialog Terminated =
provisional<BR>&gt;&nbsp;&nbsp;&nbsp;=20
response it MAY release resources and procedures associated with=20
the<BR>&gt;&nbsp;&nbsp;&nbsp; early dialog the 199 response is received=20
on.<BR>&gt;<BR>&gt; Amend to clarify the meaning of 199 independently of =
the=20
consequences<BR>&gt; of that meaning:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; =
When a=20
client receives a 199 Early Dialog Terminated=20
provisional<BR>&gt;&nbsp;&nbsp;&nbsp; response, it has positive =
knowledge that=20
the UAS that created the<BR>&gt;&nbsp;&nbsp;&nbsp; early dialog has =
terminated=20
it, and so the client MAY release<BR>&gt;&nbsp;&nbsp;&nbsp; resources =
and=20
procedures associated with the early dialog for =
which<BR>&gt;&nbsp;&nbsp;&nbsp;=20
the 199 response is received.<BR>&gt;<BR>&gt; * Section =
4<BR>&gt;<BR>&gt; Add=20
this paragraph before the last =
paragraph:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; A=20
UAC that receives a 199 response for an early dialog MUST=20
NOT<BR>&gt;&nbsp;&nbsp;&nbsp; send any further requests on that dialog, =
except=20
for a PRACK if the<BR>&gt;&nbsp;&nbsp;&nbsp; 199 response was sent =
reliably, and=20
PRACKs for other reliable<BR>&gt;&nbsp;&nbsp;&nbsp; provisional =
responses that=20
are to be acknowledged.<BR>&gt;<BR>&gt; This is copied from section 7 =
(Backward=20
Compatibility), but since it<BR>&gt; is a constraint on UACs, it should =
also be=20
stated in this section.<BR>&gt;<BR>&gt; * Section=20
4<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; The UAC MAY use additional =
information (for=20
example the final<BR>&gt;&nbsp;&nbsp;&nbsp; response which triggered the =
199=20
response) received in the 199<BR>&gt;&nbsp;&nbsp;&nbsp; response when =
initiating=20
new sessions, if it is possible to avoid the<BR>&gt;&nbsp;&nbsp;&nbsp; =
new=20
session initiation request to be rejected.<BR>&gt;<BR>&gt; What is the =
meaning=20
of the final clause?<BR>&gt;<BR>&gt; * Section 5<BR>&gt;<BR>&gt; Add =
"early" as=20
marked:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; When a UAS wants to terminate =
*an=20
early* dialog it sends a non-2xx<BR>&gt;&nbsp;&nbsp;&nbsp; SIP final =
response,=20
as specified in [RFC3261].&nbsp; In addition, =
prior<BR>&gt;&nbsp;&nbsp;&nbsp; to=20
sending the non-2xx SIP final response, the UAS MAY send a=20
199<BR>&gt;&nbsp;&nbsp;&nbsp; response to indicate that the dialog is =
being=20
terminated.<BR>&gt;<BR>&gt; This clarifies that 199 cannot be used to =
terminate=20
a confirmed<BR>&gt; dialog.<BR>&gt;<BR>&gt; (The reason a UAS may want =
to send a=20
199 is to provide this<BR>&gt; information even when the upstream proxy =
does not=20
support 199.)<BR>&gt;<BR>&gt; There is a subtlty to the second=20
sentence:<BR>&gt;<BR>&gt; Sending the 199 *before* the final response is =
not=20
necessary due to<BR>&gt; the semantics of 199 (as it will probably reach =
the UAC=20
first anyway,<BR>&gt; and it would not matter if it did not), but rather =
to=20
prevent the<BR>&gt; upstream proxy from generating a 199 based on the =
final=20
response, and<BR>&gt; then to transmit the UAS's 199 immediately after =
-- if the=20
UAS sends<BR>&gt; the 199 first, any upstream proxy will (most likely) =
see the=20
199<BR>&gt; before seeing the final response (which might induce the =
proxy to=20
send<BR>&gt; its own 199).<BR>&gt;<BR>&gt; * Section=20
6<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; When a forking proxy receives a =
non-2xx=20
final response which<BR>&gt;&nbsp;&nbsp;&nbsp; terminates one or more =
early=20
dialogs and the proxy does not intend to<BR>&gt;&nbsp;&nbsp;&nbsp; =
forward the=20
final response immediately (due to the rules for =
a<BR>&gt;&nbsp;&nbsp;&nbsp;=20
forking proxy) it MUST send a 199 provisional response, for=20
each<BR>&gt;&nbsp;&nbsp;&nbsp; associated early dialog that it can =
associate=20
with the final<BR>&gt;&nbsp;&nbsp;&nbsp; response, upstream towards the =
sender=20
of the associated request.<BR>&gt;<BR>&gt; Clarify, and note that if the =
proxy=20
has already passed a 199 for an<BR>&gt; early dialog, it need not =
generate one=20
itself.&nbsp; Also, I've reduced<BR>&gt; MUST to SHOULD, because (as the =

following NOTE explains), the proxy<BR>&gt; may not know of all early =
dialogs=20
which are terminated by the final<BR>&gt;=20
response.<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; [...] it SHOULD initiate a =
199=20
provisional response upstream<BR>&gt;&nbsp;&nbsp;&nbsp; towards the =
sender of=20
the associated request, for each terminated<BR>&gt;&nbsp;&nbsp;&nbsp; =
early=20
dialog, excepting those for which it has already=20
transmitted<BR>&gt;&nbsp;&nbsp;&nbsp; a 199 response.<BR>&gt;<BR>&gt; =
Note that=20
the "already transmitted 199 responses" may have come from<BR>&gt; =
downstream=20
proxies as well as downstream UASs, so this consideration<BR>&gt; =
applies even=20
if we do not allow UASs to send 199s.<BR>&gt;<BR>&gt; * Section=20
7<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; Since all SIP entities involved in a =
session=20
setup do not necessarily<BR>&gt;&nbsp;&nbsp;&nbsp; support the specific =
meaning=20
of the 199 Early Dialog Terminated<BR>&gt;&nbsp;&nbsp;&nbsp; provisional =

response, the sender of the response MUST be prepared=20
to<BR>&gt;&nbsp;&nbsp;&nbsp; receive SIP requests associated with the =
dialog for=20
which the 199<BR>&gt;&nbsp;&nbsp;&nbsp; response was =
sent.<BR>&gt;<BR>&gt; This=20
situation can also result due to race conditions.&nbsp; Amend=20
to:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; Because of race conditions and =
because the=20
client may not support the<BR>&gt;&nbsp;&nbsp;&nbsp; specific meaning of =
the 199=20
provisional response, the sender of the<BR>&gt;&nbsp;&nbsp;&nbsp; =
response MUST=20
be prepared to receive SIP requests associated =
with<BR>&gt;&nbsp;&nbsp;&nbsp;=20
the dialog for which the 199 response was sent.<BR>&gt;<BR>&gt; And add =
a=20
qualifier to this sentence:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; If such =
request is=20
received, and the receiver maintains state of<BR>&gt;&nbsp;&nbsp;&nbsp; =
the=20
dialogs, the receiver MUST reply to such requests with a=20
481<BR>&gt;&nbsp;&nbsp;&nbsp; final response (unless another error =
response is=20
more appropriate).<BR>&gt;<BR>&gt; * Section 7<BR>&gt;<BR>&gt; Remove =
"..." from=20
this sentence and add "even" as =
marked:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; The=20
199 Early Dialog Terminated response code MUST NOT replace=20
a<BR>&gt;&nbsp;&nbsp;&nbsp; final response.&nbsp; A final response MUST =
always=20
be sent, *even* after<BR>&gt;&nbsp;&nbsp;&nbsp; one or many 199 Early =
Dialog=20
Terminated provisional responses have<BR>&gt;&nbsp;&nbsp;&nbsp; been=20
sent.<BR>&gt;<BR>&gt; * Section 8<BR>&gt;<BR>&gt; Clarify this sentence=20
to:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; The 199 Early Dialog Terminated =
response=20
code allows a SIP entity<BR>&gt;&nbsp;&nbsp;&nbsp; to indicate to =
upstream=20
entities that a specific early dialog has<BR>&gt;&nbsp;&nbsp;&nbsp; been =

terminated, before a final response reaches the=20
upstream<BR>&gt;&nbsp;&nbsp;&nbsp; entities.<BR>&gt;<BR>&gt; Add the=20
following:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; The early dialog that has =
been=20
terminated is identified by the Call-Id<BR>&gt;&nbsp;&nbsp;&nbsp; header =
and the=20
to-tag value of the 199 response.<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; =
Future=20
standards may define optional additional information=20
carried<BR>&gt;&nbsp;&nbsp;&nbsp; in the headers and/or body of the=20
response.<BR>&gt;<BR>&gt; * Section 9<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; =
A 199=20
Early Dialog Terminated provisional response MUST NOT contain=20
a<BR>&gt;&nbsp;&nbsp;&nbsp; new SDP offer/answer message body, but the =
sender of=20
the response MAY<BR>&gt;&nbsp;&nbsp;&nbsp; insert a copy of a previously =
sent=20
offer/answer message body as<BR>&gt;&nbsp;&nbsp;&nbsp; otherwise allowed =
by the=20
offer/answer rules for a provisional<BR>&gt;&nbsp;&nbsp;&nbsp;=20
response.<BR>&gt;<BR>&gt; This rule seems excessively strict, as it =
seems OK to=20
send a new SDP<BR>&gt; answer in a 199, even if the dialog will be =
immediately=20
terminated.<BR>&gt; Certainly, the UAC cannot tell the difference, as =
there=20
could have<BR>&gt; been a previous but lost 1xx response carrying the =
same=20
SDP.<BR>&gt;<BR>&gt; The interaction of 199 and SDP probably needs =
further=20
study...<BR>&gt;<BR>&gt; * Section 11<BR>&gt;<BR>&gt; Nit:&nbsp; Use =
"reject"=20
here rather than "rejects":<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; UAS_2 and =
UAS_3=20
*reject* the INVITE by sending a 4xx error response.<BR>&gt;<BR>&gt; =
Here is a=20
modification of the example, showing how a UAS can send a<BR>&gt; 199, =
and how=20
the proxy handles it.&nbsp; (You may or may not want to use<BR>&gt; =
this.)&nbsp;=20
I've also added the ACK's for the 4xx's, which were missing in<BR>&gt; =
the=20
original.<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; [...] UAS_2 and UAS_3 reject =
the=20
INVITE by sending a 4xx error<BR>&gt;&nbsp;&nbsp;&nbsp; response.&nbsp; =
UAS_3=20
sends a 199 with its 4xx.&nbsp; When P1 receives =
the<BR>&gt;&nbsp;&nbsp;&nbsp;=20
4xx response from UAS_2 it immediately sends a 199 Early=20
Dialog<BR>&gt;&nbsp;&nbsp;&nbsp; Terminated associated with UAS_2's =
dialog=20
towards the UAC.&nbsp; P1<BR>&gt;&nbsp;&nbsp;&nbsp; transmits UAS_3's =
199, and=20
so does not initiate one itself for<BR>&gt;&nbsp;&nbsp;&nbsp; UAS_3's =
dialog=20
when it receives UAS_3's 4xx=20
response.<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
UAC&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
P1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
UAS_2&nbsp;&nbsp;&nbsp; UAS_3&nbsp;&nbsp;&nbsp;=20
UAS_4<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; --- =
INVITE=20
------&gt;<BR>&gt;&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;=20
--- INVITE (leg 2)=20
-&gt;<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;=20
--- INVITE (leg 3)=20
----------&gt;<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;=20
--- INVITE (leg 4)=20
-------------------&gt;<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&lt;-- 18x (leg 2) =
-----<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&lt;-- 18x (leg 2)=20
--<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
&lt;-- 18x (leg 3)=20
--------------<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;-- =
18x (leg=20
3)=20
--<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
&lt;-- 18x (leg 4)=20
-----------------------<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
 &lt;--=20
18x (leg 4)=20
--<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
&lt;-- 4xx (leg 2) =
-----<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&lt;-- 199 (leg 2)=20
--<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
--- ACK (leg 2)=20
----&gt;<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;=20
&lt;-- 199 (leg 3)=20
--------------<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;-- =
199 (leg=20
3)=20
--<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
&lt;-- 4xx (leg 3)=20
--------------<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;=20
--- ACK (leg 3)=20
-------------&gt;<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&lt;-- 200 (leg 4)=20
-----------------------<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
 &lt;--=20
200 (leg 4) --<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; --- ACK =
(leg 4)=20
-&gt;<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;=20
--- ACK (leg 4) ----------------------&gt;<BR>&gt;<BR>&gt; Dale<BR>&gt;=20
_______________________________________________<BR>&gt; Sip mailing =
list&nbsp;=20
<A=20
href=3D"https://www.ietf.org/mailman/listinfo/sip">https://www.ietf.org/m=
ailman/listinfo/sip</A><BR>&gt;=20
This list is for 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>&gt;<BR>_______________________________________________<BR>Sip =
mailing=20
list&nbsp; <A=20
href=3D"https://www.ietf.org/mailman/listinfo/sip">https://www.ietf.org/m=
ailman/listinfo/sip</A><BR>This=20
list is for NEW development of the core SIP Protocol<BR>Use=20
sip-implementors@cs.columbia.edu for questions on current sip<BR>Use=20
sipping@ietf.org for new developments on the application of=20
sip<BR><BR></FONT></P><BR><BR>
<P align=3Dleft><FONT style=3D"BACKGROUND-COLOR: =
#ffffff">.</FONT></P></BODY></HTML>

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_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============0688404396==--


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Thread-Topic: [Sip] Draft submission: draft-ietf-sip-199-00
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
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=20
Hi,
=20
The way the "best" final response code is chosen is described in RFC =
3261, and the intention of 199 is not to change that.
=20
Regarding whether a 199 will be sent for leg 4: if the proxy is now =
going to send a "best" final response I guess it doesn't need to send =
199 for leg 4.
=20
Regards,
=20
Christer
=20
=20
=20

________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =
ashish.hsaxena@wipro.com
Sent: 16. kes=E4kuuta 2008 10:32
To: pkyzivat@cisco.com; Dale.Worley@comcast.net
Cc: sip@ietf.org
Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00


Referring to introduction part in the draft.
   " At some point the proxy uses a specified mechanism to determine the =
"best" final response code, and forwards that final response upstream =
towards the sender of the associated request."
=20
Considering a scenario where all the forks have failed in yielding a 2xx =
response from different UASes (UAS2 , 3 & 4 in the diagram) with =
different error codes, how will the proxy determine the "best" final =
response and to which leg would it send the best final response. What =
would happen after 199 to leg 4?
=20
      UAC               P1                  UAS_2    UAS_3    UAS_4

       --- INVITE ------>
                         --- INVITE (leg 2) ->
                         --- INVITE (leg 3) ---------->
                         --- INVITE (leg 4) ------------------->
                         <-- 18x (leg 2) -----
       <-- 18x (leg 2) --
                         <-- 18x (leg 3) --------------
       <-- 18x (leg 3) --
                         <-- 18x (leg 4) -----------------------
       <-- 18x (leg 4) --
                         <-- 4xx (leg 2) -----
       <-- 199 (leg 2) --
                         <-- 5xx (leg 3) --------------
       <-- 199 (leg 3) --
                         <-- 6xx (leg 4) -----------------------
       <-- 199 (leg 4) --


Best Regards,=20
Ashish Saxena=20
877-5570=20

=20

________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =
Attila Sipos
Sent: Saturday, June 14, 2008 9:30 PM
To: Paul Kyzivat; Dale.Worley@comcast.net
Cc: sip@ietf.org
Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00




>>However I am still scratching my head about the UAS sending a 199. =
While
>>I see no reason to forbid it I'm trying to find a situation when it
>>would be helpful.

It may or may not be helpful but as you say, you can
see no reason to forbid it.

Why restrict it if the restriction isn't needed?

As someone has pointed out, a B2BUA scenario might benefit
from 199.  In fact, any kind of SIP gateway might benefit.

> This is one of those cases that could probably benefit from
> simulation to judge the merits. My gut feel is that the costs
> of the UAS sending the 199 outweigh the benefits.

If the UAS wants to send it, that is a decision for
them - not something that needs to be prevented.

SIP forking, in the technical sense of sending a
response with a different to-tag, is always legal.
So, any UA which has used forking
should also be able to use 199.

I don't think we need to include any specific text about UAS
except to say that a UAS can use it if the UAS has created
the forking.

Regards,

Attila



-----Original Message-----
From: sip-bounces@ietf.org on behalf of Paul Kyzivat
Sent: Fri 13/06/2008 15:07
To: Dale.Worley@comcast.net
Cc: sip@ietf.org
Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00

This all sounds like good stuff.

However I am still scratching my head about the UAS sending a 199. While
I see no reason to forbid it I'm trying to find a situation when it
would be helpful. The only case I have been able to think of is:

   There is a forking proxy upstream of the UAS, but it doesn't
   support 199. Yet it may delay the final response while it
   tries other forks. So the UAS sending the 199 makes up for
   the inadequacy of the forking proxy.

The trouble with this is that it is inefficient in many cases. If each
proxy that forks also supports the sending of 199, and the UAS doesn't
send one, then we avoid sending the message except in cases where it is
helpful. If the UAS always sends it we have increased the message count
by one for all failing calls. And the majority of failing calls get no
benefit from a 199.

This is one of those cases that could probably benefit from simulation
to judge the merits. My gut feel is that the costs of the UAS sending
the 199 outweigh the benefits. Barring some analysis to the contrary I
think I would say that a UAS SHOULD NOT send a 199. Or perhaps MAY send,
but without any guidance of when.

        Paul

Dale.Worley@comcast.net wrote:
> Here are some comments on this I-D.  I don't think that any of them
> modify our understanding of the proposal.  I guess this is very terse;
> please use these suggestions as you see fit.
>
> * Abstract
>
> Clarify:
>
>    This specification defines a new SIP response code, 199 Early
>    Dialog Terminated, which a SIP entity can use to indicate upstream
>    that an early dialog has been terminated.  The response code can be
>    used by a SIP entity (a SIP proxy or UAS), to inform the UAC that
>    an early dialog has been terminated.  As the 199 response is a
>    provisional response, often the UAC will receive it before
>    receiving a final response to its request.
>
> * Section 1
>
>    When a forking proxy receives a non-2xx final response, it does not
>    always immediately forward the response upstream towards the sender
>    of the associated request.  Instead, the forking proxy "stores" it
>    and waits for further final responses from remote destinations =
where
>    the forked request was forwarded.
>
> The text does not allow for serial forking, only parallel forking,
> which might be confusing to people who are more aware of serial
> forking.  So amend the last sentence to:
>
>    Instead, the forking proxy "stores" it and waits for further final
>    responses from remote destinations where the forked request was
>    forwarded, and possibly forwards the request to additional
>    destinations.
>
> Also, Avoid using "..." except when indicating that the word is used
> in a non-standard way or that one is discussing the word itself:
>
>    Instead, the forking proxy "stores" it [...]
>
> should be:
>
>    Instead, the forking proxy stores it [...]
>
> * Section 3.
>
>    REQ 1: It must be possible to indicate to the UAC that an early
>    dialog has been terminated before a final response is sent.
>
> Oddly, though everyone thinks they understand this sentence, it
> doesn't really say what we want.  To really ensure that the UAC
> understands the early dialog has been terminated before the UAS sends
> a final response, the UAS would have to send 199 *and get the PRACK
> saying that it was received* before sending a final response.  This =
may
> well be worse than the present state of affairs.
>
> What we really mean is:
>
>    REQ 1: It must be possible for a UAS or proxy to send an indication
>    to the UAC that an early dialog has been terminated, which
>    indication will not be delayed waiting for a final response from
>    any UAS.
>
> * Section 4
>
>    When a client receives a 199 Early Dialog Terminated provisional
>    response it MAY release resources and procedures associated with =
the
>    early dialog the 199 response is received on.
>
> Amend to clarify the meaning of 199 independently of the consequences
> of that meaning:
>
>    When a client receives a 199 Early Dialog Terminated provisional
>    response, it has positive knowledge that the UAS that created the
>    early dialog has terminated it, and so the client MAY release
>    resources and procedures associated with the early dialog for which
>    the 199 response is received.
>
> * Section 4
>
> Add this paragraph before the last paragraph:
>
>    A UAC that receives a 199 response for an early dialog MUST NOT
>    send any further requests on that dialog, except for a PRACK if the
>    199 response was sent reliably, and PRACKs for other reliable
>    provisional responses that are to be acknowledged.
>
> This is copied from section 7 (Backward Compatibility), but since it
> is a constraint on UACs, it should also be stated in this section.
>
> * Section 4
>
>    The UAC MAY use additional information (for example the final
>    response which triggered the 199 response) received in the 199
>    response when initiating new sessions, if it is possible to avoid =
the
>    new session initiation request to be rejected.
>
> What is the meaning of the final clause?
>
> * Section 5
>
> Add "early" as marked:
>
>    When a UAS wants to terminate *an early* dialog it sends a non-2xx
>    SIP final response, as specified in [RFC3261].  In addition, prior
>    to sending the non-2xx SIP final response, the UAS MAY send a 199
>    response to indicate that the dialog is being terminated.
>
> This clarifies that 199 cannot be used to terminate a confirmed
> dialog.
>
> (The reason a UAS may want to send a 199 is to provide this
> information even when the upstream proxy does not support 199.)
>
> There is a subtlty to the second sentence:
>
> Sending the 199 *before* the final response is not necessary due to
> the semantics of 199 (as it will probably reach the UAC first anyway,
> and it would not matter if it did not), but rather to prevent the
> upstream proxy from generating a 199 based on the final response, and
> then to transmit the UAS's 199 immediately after -- if the UAS sends
> the 199 first, any upstream proxy will (most likely) see the 199
> before seeing the final response (which might induce the proxy to send
> its own 199).
>
> * Section 6
>
>    When a forking proxy receives a non-2xx final response which
>    terminates one or more early dialogs and the proxy does not intend =
to
>    forward the final response immediately (due to the rules for a
>    forking proxy) it MUST send a 199 provisional response, for each
>    associated early dialog that it can associate with the final
>    response, upstream towards the sender of the associated request.
>
> Clarify, and note that if the proxy has already passed a 199 for an
> early dialog, it need not generate one itself.  Also, I've reduced
> MUST to SHOULD, because (as the following NOTE explains), the proxy
> may not know of all early dialogs which are terminated by the final
> response.
>
>    [...] it SHOULD initiate a 199 provisional response upstream
>    towards the sender of the associated request, for each terminated
>    early dialog, excepting those for which it has already transmitted
>    a 199 response.
>
> Note that the "already transmitted 199 responses" may have come from
> downstream proxies as well as downstream UASs, so this consideration
> applies even if we do not allow UASs to send 199s.
>
> * Section 7
>
>    Since all SIP entities involved in a session setup do not =
necessarily
>    support the specific meaning of the 199 Early Dialog Terminated
>    provisional response, the sender of the response MUST be prepared =
to
>    receive SIP requests associated with the dialog for which the 199
>    response was sent.
>
> This situation can also result due to race conditions.  Amend to:
>
>    Because of race conditions and because the client may not support =
the
>    specific meaning of the 199 provisional response, the sender of the
>    response MUST be prepared to receive SIP requests associated with
>    the dialog for which the 199 response was sent.
>
> And add a qualifier to this sentence:
>
>    If such request is received, and the receiver maintains state of
>    the dialogs, the receiver MUST reply to such requests with a 481
>    final response (unless another error response is more appropriate).
>
> * Section 7
>
> Remove "..." from this sentence and add "even" as marked:
>
>    The 199 Early Dialog Terminated response code MUST NOT replace a
>    final response.  A final response MUST always be sent, *even* after
>    one or many 199 Early Dialog Terminated provisional responses have
>    been sent.
>
> * Section 8
>
> Clarify this sentence to:
>
>    The 199 Early Dialog Terminated response code allows a SIP entity
>    to indicate to upstream entities that a specific early dialog has
>    been terminated, before a final response reaches the upstream
>    entities.
>
> Add the following:
>
>    The early dialog that has been terminated is identified by the =
Call-Id
>    header and the to-tag value of the 199 response.
>
>    Future standards may define optional additional information carried
>    in the headers and/or body of the response.
>
> * Section 9
>
>    A 199 Early Dialog Terminated provisional response MUST NOT contain =
a
>    new SDP offer/answer message body, but the sender of the response =
MAY
>    insert a copy of a previously sent offer/answer message body as
>    otherwise allowed by the offer/answer rules for a provisional
>    response.
>
> This rule seems excessively strict, as it seems OK to send a new SDP
> answer in a 199, even if the dialog will be immediately terminated.
> Certainly, the UAC cannot tell the difference, as there could have
> been a previous but lost 1xx response carrying the same SDP.
>
> The interaction of 199 and SDP probably needs further study...
>
> * Section 11
>
> Nit:  Use "reject" here rather than "rejects":
>
>    UAS_2 and UAS_3 *reject* the INVITE by sending a 4xx error =
response.
>
> Here is a modification of the example, showing how a UAS can send a
> 199, and how the proxy handles it.  (You may or may not want to use
> this.)  I've also added the ACK's for the 4xx's, which were missing in
> the original.
>
>    [...] UAS_2 and UAS_3 reject the INVITE by sending a 4xx error
>    response.  UAS_3 sends a 199 with its 4xx.  When P1 receives the
>    4xx response from UAS_2 it immediately sends a 199 Early Dialog
>    Terminated associated with UAS_2's dialog towards the UAC.  P1
>    transmits UAS_3's 199, and so does not initiate one itself for
>    UAS_3's dialog when it receives UAS_3's 4xx response.
>
>       UAC               P1                  UAS_2    UAS_3    UAS_4
>
>        --- INVITE ------>
>                          --- INVITE (leg 2) ->
>                          --- INVITE (leg 3) ---------->
>                          --- INVITE (leg 4) ------------------->
>                          <-- 18x (leg 2) -----
>        <-- 18x (leg 2) --
>                          <-- 18x (leg 3) --------------
>        <-- 18x (leg 3) --
>                          <-- 18x (leg 4) -----------------------
>        <-- 18x (leg 4) --
>                          <-- 4xx (leg 2) -----
>        <-- 199 (leg 2) --
>                          --- ACK (leg 2) ---->
>                          <-- 199 (leg 3) --------------
>        <-- 199 (leg 3) --
>                          <-- 4xx (leg 3) --------------
>                          --- ACK (leg 3) ------------->
>                          <-- 200 (leg 4) -----------------------
>        <-- 200 (leg 4) --
>        --- ACK (leg 4) ->
>                          --- ACK (leg 4) ---------------------->
>
> Dale
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip





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<HTML><HEAD><TITLE>RE: [Sip] Draft submission: =
draft-ietf-sip-199-00</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2800.1609" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D473410708-16062008>Hi,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D473410708-16062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D473410708-16062008>The=20
way the "best" final response code is chosen is described in RFC 3261, =
and the=20
intention of 199 is not to change that.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D473410708-16062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D473410708-16062008>Regarding whether a 199 will be sent for leg =
4: if the=20
proxy is now going to send a "best" final response I guess it doesn't =
need to=20
send 199 for leg 4.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D473410708-16062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D473410708-16062008>Regards,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D473410708-16062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D473410708-16062008>Christer</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D473410708-16062008></SPAN></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>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
[mailto:sip-bounces@ietf.org] <B>On Behalf Of=20
</B>ashish.hsaxena@wipro.com<BR><B>Sent:</B> 16. kes=E4kuuta 2008=20
10:32<BR><B>To:</B> pkyzivat@cisco.com; =
Dale.Worley@comcast.net<BR><B>Cc:</B>=20
sip@ietf.org<BR><B>Subject:</B> Re: [Sip] Draft submission:=20
draft-ietf-sip-199-00<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV><SPAN class=3D937291707-16062008><FONT face=3DVerdana =
color=3D#800000=20
size=3D2>Referring to introduction part in the =
draft.</FONT></SPAN></DIV>
<DIV><SPAN class=3D937291707-16062008>&nbsp;&nbsp;&nbsp;" At some point =
the proxy=20
uses a&nbsp;specified mechanism to determine the "best" final response =
code,=20
and&nbsp;forwards that final response upstream towards the sender of=20
the&nbsp;associated request."</SPAN></DIV>
<DIV><SPAN class=3D937291707-16062008><FONT face=3DVerdana =
color=3D#800000=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D937291707-16062008><FONT face=3DVerdana =
color=3D#800000=20
size=3D2>Considering a scenario where all the forks have failed in =
yielding a 2xx=20
response from different UASes (UAS2 , 3 &amp; 4 in the diagram) with =
different=20
error codes, how will the proxy determine the "best" final response and =
to which=20
leg would it send the best final response. What would happen after 199 =
to leg=20
4?</FONT></SPAN></DIV>
<DIV><SPAN class=3D937291707-16062008><FONT face=3DVerdana =
color=3D#800000=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D937291707-16062008>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
UAC&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
P1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
UAS_2&nbsp;&nbsp;&nbsp; UAS_3&nbsp;&nbsp;&nbsp;=20
UAS_4<BR><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; --- INVITE=20
------&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;=20
--- INVITE (leg 2)=20
-&gt;<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;=20
--- INVITE (leg 3)=20
----------&gt;<BR>&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;=20
--- INVITE (leg 4)=20
-------------------&gt;<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;=20
&lt;-- 18x (leg 2) -----<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;-- =
18x (leg=20
2)=20
--<BR>&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;=20
&lt;-- 18x (leg 3) =
--------------<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;--=20
18x (leg 3)=20
--<BR>&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;=20
&lt;-- 18x (leg 4)=20
-----------------------<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;-- =
18x (leg=20
4)=20
--<BR>&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;=20
&lt;-- <STRONG>4xx</STRONG> (leg 2)=20
-----<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;-- 199 (leg 2)=20
--<BR>&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;=20
&lt;-- <STRONG>5xx </STRONG>(leg 3)=20
--------------<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;-- 199 (leg =
3)=20
--<BR>&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;=20
&lt;--&nbsp;<STRONG>6xx </STRONG>(leg 4)=20
-----------------------<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&lt;--&nbsp;199=20
(leg 4) --<BR></SPAN></DIV><!-- Converted from text/rtf format -->
<P><SPAN lang=3Den-us><FONT face=3DVerdana color=3D#800000 size=3D2>Best =

Regards,</FONT></SPAN> <BR><SPAN lang=3Den-us><FONT face=3DVerdana =
color=3D#800000=20
size=3D2>Ashish Saxena</FONT></SPAN> <BR><SPAN lang=3Den-us><FONT =
face=3DVerdana=20
color=3D#800000 size=3D2>877-5570</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> sip-bounces@ietf.org=20
[mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Attila =
Sipos<BR><B>Sent:</B>=20
Saturday, June 14, 2008 9:30 PM<BR><B>To:</B> Paul Kyzivat;=20
Dale.Worley@comcast.net<BR><B>Cc:</B> sip@ietf.org<BR><B>Subject:</B> =
Re: [Sip]=20
Draft submission: draft-ietf-sip-199-00<BR></FONT><BR></DIV>
<DIV></DIV><!-- Converted from text/plain format --><BR>
<P><FONT size=3D2>&gt;&gt;However I am still scratching my head about =
the UAS=20
sending a 199. While<BR>&gt;&gt;I see no reason to forbid it I'm trying =
to find=20
a situation when it<BR>&gt;&gt;would be helpful.<BR><BR>It may or may =
not be=20
helpful but as you say, you can<BR>see no reason to forbid =
it.<BR><BR>Why=20
restrict it if the restriction isn't needed?<BR><BR>As someone has =
pointed out,=20
a B2BUA scenario might benefit<BR>from 199.&nbsp; In fact, any kind of =
SIP=20
gateway might benefit.<BR><BR>&gt; This is one of those cases that could =

probably benefit from<BR>&gt; simulation to judge the merits. My gut =
feel is=20
that the costs<BR>&gt; of the UAS sending the 199 outweigh the=20
benefits.<BR><BR>If the UAS wants to send it, that is a decision =
for<BR>them -=20
not something that needs to be prevented.<BR><BR>SIP forking, in the =
technical=20
sense of sending a<BR>response with a different to-tag, is always =
legal.<BR>So,=20
any UA which has used forking<BR>should also be able to use =
199.<BR><BR>I don't=20
think we need to include any specific text about UAS<BR>except to say =
that a UAS=20
can use it if the UAS has created<BR>the=20
forking.<BR><BR>Regards,<BR><BR>Attila<BR><BR><BR><BR>-----Original=20
Message-----<BR>From: sip-bounces@ietf.org on behalf of Paul =
Kyzivat<BR>Sent:=20
Fri 13/06/2008 15:07<BR>To: Dale.Worley@comcast.net<BR>Cc:=20
sip@ietf.org<BR>Subject: Re: [Sip] Draft submission:=20
draft-ietf-sip-199-00<BR><BR>This all sounds like good =
stuff.<BR><BR>However I=20
am still scratching my head about the UAS sending a 199. While<BR>I see =
no=20
reason to forbid it I'm trying to find a situation when it<BR>would be =
helpful.=20
The only case I have been able to think of is:<BR><BR>&nbsp;&nbsp; There =
is a=20
forking proxy upstream of the UAS, but it doesn't<BR>&nbsp;&nbsp; =
support 199.=20
Yet it may delay the final response while it<BR>&nbsp;&nbsp; tries other =
forks.=20
So the UAS sending the 199 makes up for<BR>&nbsp;&nbsp; the inadequacy =
of the=20
forking proxy.<BR><BR>The trouble with this is that it is inefficient in =
many=20
cases. If each<BR>proxy that forks also supports the sending of 199, and =
the UAS=20
doesn't<BR>send one, then we avoid sending the message except in cases =
where it=20
is<BR>helpful. If the UAS always sends it we have increased the message=20
count<BR>by one for all failing calls. And the majority of failing calls =
get=20
no<BR>benefit from a 199.<BR><BR>This is one of those cases that could =
probably=20
benefit from simulation<BR>to judge the merits. My gut feel is that the =
costs of=20
the UAS sending<BR>the 199 outweigh the benefits. Barring some analysis =
to the=20
contrary I<BR>think I would say that a UAS SHOULD NOT send a 199. Or =
perhaps MAY=20
send,<BR>but without any guidance of=20
when.<BR><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
Paul<BR><BR>Dale.Worley@comcast.net wrote:<BR>&gt; Here are some =
comments on=20
this I-D.&nbsp; I don't think that any of them<BR>&gt; modify our =
understanding=20
of the proposal.&nbsp; I guess this is very terse;<BR>&gt; please use =
these=20
suggestions as you see fit.<BR>&gt;<BR>&gt; * Abstract<BR>&gt;<BR>&gt;=20
Clarify:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; This specification defines a =
new SIP=20
response code, 199 Early<BR>&gt;&nbsp;&nbsp;&nbsp; Dialog Terminated, =
which a=20
SIP entity can use to indicate upstream<BR>&gt;&nbsp;&nbsp;&nbsp; that =
an early=20
dialog has been terminated.&nbsp; The response code can=20
be<BR>&gt;&nbsp;&nbsp;&nbsp; used by a SIP entity (a SIP proxy or UAS), =
to=20
inform the UAC that<BR>&gt;&nbsp;&nbsp;&nbsp; an early dialog has been=20
terminated.&nbsp; As the 199 response is a<BR>&gt;&nbsp;&nbsp;&nbsp; =
provisional=20
response, often the UAC will receive it before<BR>&gt;&nbsp;&nbsp;&nbsp; =

receiving a final response to its request.<BR>&gt;<BR>&gt; * Section=20
1<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; When a forking proxy receives a =
non-2xx=20
final response, it does not<BR>&gt;&nbsp;&nbsp;&nbsp; always immediately =
forward=20
the response upstream towards the sender<BR>&gt;&nbsp;&nbsp;&nbsp; of =
the=20
associated request.&nbsp; Instead, the forking proxy "stores"=20
it<BR>&gt;&nbsp;&nbsp;&nbsp; and waits for further final responses from =
remote=20
destinations where<BR>&gt;&nbsp;&nbsp;&nbsp; the forked request was=20
forwarded.<BR>&gt;<BR>&gt; The text does not allow for serial forking, =
only=20
parallel forking,<BR>&gt; which might be confusing to people who are =
more aware=20
of serial<BR>&gt; forking.&nbsp; So amend the last sentence=20
to:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; Instead, the forking proxy =
"stores" it and=20
waits for further final<BR>&gt;&nbsp;&nbsp;&nbsp; responses from remote=20
destinations where the forked request was<BR>&gt;&nbsp;&nbsp;&nbsp; =
forwarded,=20
and possibly forwards the request to =
additional<BR>&gt;&nbsp;&nbsp;&nbsp;=20
destinations.<BR>&gt;<BR>&gt; Also, Avoid using "..." except when =
indicating=20
that the word is used<BR>&gt; in a non-standard way or that one is =
discussing=20
the word itself:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; Instead, the forking =
proxy=20
"stores" it [...]<BR>&gt;<BR>&gt; should =
be:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp;=20
Instead, the forking proxy stores it [...]<BR>&gt;<BR>&gt; * Section=20
3.<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; REQ 1: It must be possible to =
indicate to=20
the UAC that an early<BR>&gt;&nbsp;&nbsp;&nbsp; dialog has been =
terminated=20
before a final response is sent.<BR>&gt;<BR>&gt; Oddly, though everyone =
thinks=20
they understand this sentence, it<BR>&gt; doesn't really say what we =
want.&nbsp;=20
To really ensure that the UAC<BR>&gt; understands the early dialog has =
been=20
terminated before the UAS sends<BR>&gt; a final response, the UAS would =
have to=20
send 199 *and get the PRACK<BR>&gt; saying that it was received* before =
sending=20
a final response.&nbsp; This may<BR>&gt; well be worse than the present =
state of=20
affairs.<BR>&gt;<BR>&gt; What we really mean=20
is:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; REQ 1: It must be possible for a =
UAS or=20
proxy to send an indication<BR>&gt;&nbsp;&nbsp;&nbsp; to the UAC that an =
early=20
dialog has been terminated, which<BR>&gt;&nbsp;&nbsp;&nbsp; indication =
will not=20
be delayed waiting for a final response from<BR>&gt;&nbsp;&nbsp;&nbsp; =
any=20
UAS.<BR>&gt;<BR>&gt; * Section 4<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; When =
a client=20
receives a 199 Early Dialog Terminated =
provisional<BR>&gt;&nbsp;&nbsp;&nbsp;=20
response it MAY release resources and procedures associated with=20
the<BR>&gt;&nbsp;&nbsp;&nbsp; early dialog the 199 response is received=20
on.<BR>&gt;<BR>&gt; Amend to clarify the meaning of 199 independently of =
the=20
consequences<BR>&gt; of that meaning:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; =
When a=20
client receives a 199 Early Dialog Terminated=20
provisional<BR>&gt;&nbsp;&nbsp;&nbsp; response, it has positive =
knowledge that=20
the UAS that created the<BR>&gt;&nbsp;&nbsp;&nbsp; early dialog has =
terminated=20
it, and so the client MAY release<BR>&gt;&nbsp;&nbsp;&nbsp; resources =
and=20
procedures associated with the early dialog for =
which<BR>&gt;&nbsp;&nbsp;&nbsp;=20
the 199 response is received.<BR>&gt;<BR>&gt; * Section =
4<BR>&gt;<BR>&gt; Add=20
this paragraph before the last =
paragraph:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; A=20
UAC that receives a 199 response for an early dialog MUST=20
NOT<BR>&gt;&nbsp;&nbsp;&nbsp; send any further requests on that dialog, =
except=20
for a PRACK if the<BR>&gt;&nbsp;&nbsp;&nbsp; 199 response was sent =
reliably, and=20
PRACKs for other reliable<BR>&gt;&nbsp;&nbsp;&nbsp; provisional =
responses that=20
are to be acknowledged.<BR>&gt;<BR>&gt; This is copied from section 7 =
(Backward=20
Compatibility), but since it<BR>&gt; is a constraint on UACs, it should =
also be=20
stated in this section.<BR>&gt;<BR>&gt; * Section=20
4<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; The UAC MAY use additional =
information (for=20
example the final<BR>&gt;&nbsp;&nbsp;&nbsp; response which triggered the =
199=20
response) received in the 199<BR>&gt;&nbsp;&nbsp;&nbsp; response when =
initiating=20
new sessions, if it is possible to avoid the<BR>&gt;&nbsp;&nbsp;&nbsp; =
new=20
session initiation request to be rejected.<BR>&gt;<BR>&gt; What is the =
meaning=20
of the final clause?<BR>&gt;<BR>&gt; * Section 5<BR>&gt;<BR>&gt; Add =
"early" as=20
marked:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; When a UAS wants to terminate =
*an=20
early* dialog it sends a non-2xx<BR>&gt;&nbsp;&nbsp;&nbsp; SIP final =
response,=20
as specified in [RFC3261].&nbsp; In addition, =
prior<BR>&gt;&nbsp;&nbsp;&nbsp; to=20
sending the non-2xx SIP final response, the UAS MAY send a=20
199<BR>&gt;&nbsp;&nbsp;&nbsp; response to indicate that the dialog is =
being=20
terminated.<BR>&gt;<BR>&gt; This clarifies that 199 cannot be used to =
terminate=20
a confirmed<BR>&gt; dialog.<BR>&gt;<BR>&gt; (The reason a UAS may want =
to send a=20
199 is to provide this<BR>&gt; information even when the upstream proxy =
does not=20
support 199.)<BR>&gt;<BR>&gt; There is a subtlty to the second=20
sentence:<BR>&gt;<BR>&gt; Sending the 199 *before* the final response is =
not=20
necessary due to<BR>&gt; the semantics of 199 (as it will probably reach =
the UAC=20
first anyway,<BR>&gt; and it would not matter if it did not), but rather =
to=20
prevent the<BR>&gt; upstream proxy from generating a 199 based on the =
final=20
response, and<BR>&gt; then to transmit the UAS's 199 immediately after =
-- if the=20
UAS sends<BR>&gt; the 199 first, any upstream proxy will (most likely) =
see the=20
199<BR>&gt; before seeing the final response (which might induce the =
proxy to=20
send<BR>&gt; its own 199).<BR>&gt;<BR>&gt; * Section=20
6<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; When a forking proxy receives a =
non-2xx=20
final response which<BR>&gt;&nbsp;&nbsp;&nbsp; terminates one or more =
early=20
dialogs and the proxy does not intend to<BR>&gt;&nbsp;&nbsp;&nbsp; =
forward the=20
final response immediately (due to the rules for =
a<BR>&gt;&nbsp;&nbsp;&nbsp;=20
forking proxy) it MUST send a 199 provisional response, for=20
each<BR>&gt;&nbsp;&nbsp;&nbsp; associated early dialog that it can =
associate=20
with the final<BR>&gt;&nbsp;&nbsp;&nbsp; response, upstream towards the =
sender=20
of the associated request.<BR>&gt;<BR>&gt; Clarify, and note that if the =
proxy=20
has already passed a 199 for an<BR>&gt; early dialog, it need not =
generate one=20
itself.&nbsp; Also, I've reduced<BR>&gt; MUST to SHOULD, because (as the =

following NOTE explains), the proxy<BR>&gt; may not know of all early =
dialogs=20
which are terminated by the final<BR>&gt;=20
response.<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; [...] it SHOULD initiate a =
199=20
provisional response upstream<BR>&gt;&nbsp;&nbsp;&nbsp; towards the =
sender of=20
the associated request, for each terminated<BR>&gt;&nbsp;&nbsp;&nbsp; =
early=20
dialog, excepting those for which it has already=20
transmitted<BR>&gt;&nbsp;&nbsp;&nbsp; a 199 response.<BR>&gt;<BR>&gt; =
Note that=20
the "already transmitted 199 responses" may have come from<BR>&gt; =
downstream=20
proxies as well as downstream UASs, so this consideration<BR>&gt; =
applies even=20
if we do not allow UASs to send 199s.<BR>&gt;<BR>&gt; * Section=20
7<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; Since all SIP entities involved in a =
session=20
setup do not necessarily<BR>&gt;&nbsp;&nbsp;&nbsp; support the specific =
meaning=20
of the 199 Early Dialog Terminated<BR>&gt;&nbsp;&nbsp;&nbsp; provisional =

response, the sender of the response MUST be prepared=20
to<BR>&gt;&nbsp;&nbsp;&nbsp; receive SIP requests associated with the =
dialog for=20
which the 199<BR>&gt;&nbsp;&nbsp;&nbsp; response was =
sent.<BR>&gt;<BR>&gt; This=20
situation can also result due to race conditions.&nbsp; Amend=20
to:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; Because of race conditions and =
because the=20
client may not support the<BR>&gt;&nbsp;&nbsp;&nbsp; specific meaning of =
the 199=20
provisional response, the sender of the<BR>&gt;&nbsp;&nbsp;&nbsp; =
response MUST=20
be prepared to receive SIP requests associated =
with<BR>&gt;&nbsp;&nbsp;&nbsp;=20
the dialog for which the 199 response was sent.<BR>&gt;<BR>&gt; And add =
a=20
qualifier to this sentence:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; If such =
request is=20
received, and the receiver maintains state of<BR>&gt;&nbsp;&nbsp;&nbsp; =
the=20
dialogs, the receiver MUST reply to such requests with a=20
481<BR>&gt;&nbsp;&nbsp;&nbsp; final response (unless another error =
response is=20
more appropriate).<BR>&gt;<BR>&gt; * Section 7<BR>&gt;<BR>&gt; Remove =
"..." from=20
this sentence and add "even" as =
marked:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; The=20
199 Early Dialog Terminated response code MUST NOT replace=20
a<BR>&gt;&nbsp;&nbsp;&nbsp; final response.&nbsp; A final response MUST =
always=20
be sent, *even* after<BR>&gt;&nbsp;&nbsp;&nbsp; one or many 199 Early =
Dialog=20
Terminated provisional responses have<BR>&gt;&nbsp;&nbsp;&nbsp; been=20
sent.<BR>&gt;<BR>&gt; * Section 8<BR>&gt;<BR>&gt; Clarify this sentence=20
to:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; The 199 Early Dialog Terminated =
response=20
code allows a SIP entity<BR>&gt;&nbsp;&nbsp;&nbsp; to indicate to =
upstream=20
entities that a specific early dialog has<BR>&gt;&nbsp;&nbsp;&nbsp; been =

terminated, before a final response reaches the=20
upstream<BR>&gt;&nbsp;&nbsp;&nbsp; entities.<BR>&gt;<BR>&gt; Add the=20
following:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; The early dialog that has =
been=20
terminated is identified by the Call-Id<BR>&gt;&nbsp;&nbsp;&nbsp; header =
and the=20
to-tag value of the 199 response.<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; =
Future=20
standards may define optional additional information=20
carried<BR>&gt;&nbsp;&nbsp;&nbsp; in the headers and/or body of the=20
response.<BR>&gt;<BR>&gt; * Section 9<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; =
A 199=20
Early Dialog Terminated provisional response MUST NOT contain=20
a<BR>&gt;&nbsp;&nbsp;&nbsp; new SDP offer/answer message body, but the =
sender of=20
the response MAY<BR>&gt;&nbsp;&nbsp;&nbsp; insert a copy of a previously =
sent=20
offer/answer message body as<BR>&gt;&nbsp;&nbsp;&nbsp; otherwise allowed =
by the=20
offer/answer rules for a provisional<BR>&gt;&nbsp;&nbsp;&nbsp;=20
response.<BR>&gt;<BR>&gt; This rule seems excessively strict, as it =
seems OK to=20
send a new SDP<BR>&gt; answer in a 199, even if the dialog will be =
immediately=20
terminated.<BR>&gt; Certainly, the UAC cannot tell the difference, as =
there=20
could have<BR>&gt; been a previous but lost 1xx response carrying the =
same=20
SDP.<BR>&gt;<BR>&gt; The interaction of 199 and SDP probably needs =
further=20
study...<BR>&gt;<BR>&gt; * Section 11<BR>&gt;<BR>&gt; Nit:&nbsp; Use =
"reject"=20
here rather than "rejects":<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; UAS_2 and =
UAS_3=20
*reject* the INVITE by sending a 4xx error response.<BR>&gt;<BR>&gt; =
Here is a=20
modification of the example, showing how a UAS can send a<BR>&gt; 199, =
and how=20
the proxy handles it.&nbsp; (You may or may not want to use<BR>&gt; =
this.)&nbsp;=20
I've also added the ACK's for the 4xx's, which were missing in<BR>&gt; =
the=20
original.<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; [...] UAS_2 and UAS_3 reject =
the=20
INVITE by sending a 4xx error<BR>&gt;&nbsp;&nbsp;&nbsp; response.&nbsp; =
UAS_3=20
sends a 199 with its 4xx.&nbsp; When P1 receives =
the<BR>&gt;&nbsp;&nbsp;&nbsp;=20
4xx response from UAS_2 it immediately sends a 199 Early=20
Dialog<BR>&gt;&nbsp;&nbsp;&nbsp; Terminated associated with UAS_2's =
dialog=20
towards the UAC.&nbsp; P1<BR>&gt;&nbsp;&nbsp;&nbsp; transmits UAS_3's =
199, and=20
so does not initiate one itself for<BR>&gt;&nbsp;&nbsp;&nbsp; UAS_3's =
dialog=20
when it receives UAS_3's 4xx=20
response.<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
UAC&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
P1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
UAS_2&nbsp;&nbsp;&nbsp; UAS_3&nbsp;&nbsp;&nbsp;=20
UAS_4<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; --- =
INVITE=20
------&gt;<BR>&gt;&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;=20
--- INVITE (leg 2)=20
-&gt;<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;=20
--- INVITE (leg 3)=20
----------&gt;<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;=20
--- INVITE (leg 4)=20
-------------------&gt;<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&lt;-- 18x (leg 2) =
-----<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&lt;-- 18x (leg 2)=20
--<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
&lt;-- 18x (leg 3)=20
--------------<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;-- =
18x (leg=20
3)=20
--<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
&lt;-- 18x (leg 4)=20
-----------------------<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
 &lt;--=20
18x (leg 4)=20
--<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
&lt;-- 4xx (leg 2) =
-----<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&lt;-- 199 (leg 2)=20
--<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
--- ACK (leg 2)=20
----&gt;<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;=20
&lt;-- 199 (leg 3)=20
--------------<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;-- =
199 (leg=20
3)=20
--<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
&lt;-- 4xx (leg 3)=20
--------------<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;=20
--- ACK (leg 3)=20
-------------&gt;<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&lt;-- 200 (leg 4)=20
-----------------------<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
 &lt;--=20
200 (leg 4) --<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; --- ACK =
(leg 4)=20
-&gt;<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;=20
--- ACK (leg 4) ----------------------&gt;<BR>&gt;<BR>&gt; Dale<BR>&gt;=20
_______________________________________________<BR>&gt; Sip mailing =
list&nbsp;=20
<A=20
href=3D"https://www.ietf.org/mailman/listinfo/sip">https://www.ietf.org/m=
ailman/listinfo/sip</A><BR>&gt;=20
This list is for 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>&gt;<BR>_______________________________________________<BR>Sip =
mailing=20
list&nbsp; <A=20
href=3D"https://www.ietf.org/mailman/listinfo/sip">https://www.ietf.org/m=
ailman/listinfo/sip</A><BR>This=20
list is for NEW development of the core SIP Protocol<BR>Use=20
sip-implementors@cs.columbia.edu for questions on current sip<BR>Use=20
sipping@ietf.org for new developments on the application of=20
sip<BR><BR></FONT></P><BR><BR>
<P align=3Dleft><FONT style=3D"BACKGROUND-COLOR: =
#ffffff">.</FONT></P></BODY></HTML>

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_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============2045265131==--


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Thread-Topic: [Sip] Draft submission: draft-ietf-sip-199-00
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From: <sreerekha.shenoy@wipro.com>
To: <christer.holmberg@ericsson.com>, <ashish.hsaxena@wipro.com>,
	<pkyzivat@cisco.com>, <Dale.Worley@comcast.net>
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Thanks Christer,
We have a query on our basic understanding of this draft.
=20
I too had the same query as Ashish, I can rephrase it for all of us.
What we mean to ask is that, considering the scenario shown by the SIP =
msg flow below, at what point does the proxy decide to send 199 response =
and on which call legs.
=20
For e.g. in the scenario below, when proxy receives 403 on leg 2, and it =
is yet to receive responses on leg 3 and 4. In this case, the proxy =
cannot send a 199 to the UAC (is my understanding, please correct if =
wrong).
Also after having received 5xx and 6xx on the leg 3 and 4, there is no =
use sending 199 now because the best final response sent to UAC is =
sufficient to indicate that the leg 3 and 4 are terminated.
=20
=20
Please correct us if we are missing some hidden lines in the draft w.r.t =
our understanding.
=20
Thanks,
Best Regards,
Rekha
(ESN : 6-877-5834)
(PSTN : 81168)
=20

________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =
Christer Holmberg
Sent: Monday, June 16, 2008 1:42 PM
To: Ashish Saxena (WT01 - Telecom Equipment); pkyzivat@cisco.com; =
Dale.Worley@comcast.net
Cc: sip@ietf.org
Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00


=20
Hi,
=20
The way the "best" final response code is chosen is described in RFC =
3261, and the intention of 199 is not to change that.
=20
Regarding whether a 199 will be sent for leg 4: if the proxy is now =
going to send a "best" final response I guess it doesn't need to send =
199 for leg 4.
=20
Regards,
=20
Christer
=20
=20
=20

________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =
ashish.hsaxena@wipro.com
Sent: 16. kes=E4kuuta 2008 10:32
To: pkyzivat@cisco.com; Dale.Worley@comcast.net
Cc: sip@ietf.org
Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00


Referring to introduction part in the draft.
   " At some point the proxy uses a specified mechanism to determine the =
"best" final response code, and forwards that final response upstream =
towards the sender of the associated request."
=20
Considering a scenario where all the forks have failed in yielding a 2xx =
response from different UASes (UAS2 , 3 & 4 in the diagram) with =
different error codes, how will the proxy determine the "best" final =
response and to which leg would it send the best final response. What =
would happen after 199 to leg 4?
=20
      UAC               P1                  UAS_2    UAS_3    UAS_4

       --- INVITE ------>
                         --- INVITE (leg 2) ->
                         --- INVITE (leg 3) ---------->
                         --- INVITE (leg 4) ------------------->
                         <-- 18x (leg 2) -----
       <-- 18x (leg 2) --
                         <-- 18x (leg 3) --------------
       <-- 18x (leg 3) --
                         <-- 18x (leg 4) -----------------------
       <-- 18x (leg 4) --
                         <-- 4xx (leg 2) -----
       <-- 199 (leg 2) --
                         <-- 5xx (leg 3) --------------
       <-- 199 (leg 3) --
                         <-- 6xx (leg 4) -----------------------
       <-- 199 (leg 4) --


Best Regards,=20
Ashish Saxena=20
877-5570=20

=20

________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =
Attila Sipos
Sent: Saturday, June 14, 2008 9:30 PM
To: Paul Kyzivat; Dale.Worley@comcast.net
Cc: sip@ietf.org
Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00




>>However I am still scratching my head about the UAS sending a 199. =
While
>>I see no reason to forbid it I'm trying to find a situation when it
>>would be helpful.

It may or may not be helpful but as you say, you can
see no reason to forbid it.

Why restrict it if the restriction isn't needed?

As someone has pointed out, a B2BUA scenario might benefit
from 199.  In fact, any kind of SIP gateway might benefit.

> This is one of those cases that could probably benefit from
> simulation to judge the merits. My gut feel is that the costs
> of the UAS sending the 199 outweigh the benefits.

If the UAS wants to send it, that is a decision for
them - not something that needs to be prevented.

SIP forking, in the technical sense of sending a
response with a different to-tag, is always legal.
So, any UA which has used forking
should also be able to use 199.

I don't think we need to include any specific text about UAS
except to say that a UAS can use it if the UAS has created
the forking.

Regards,

Attila



-----Original Message-----
From: sip-bounces@ietf.org on behalf of Paul Kyzivat
Sent: Fri 13/06/2008 15:07
To: Dale.Worley@comcast.net
Cc: sip@ietf.org
Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00

This all sounds like good stuff.

However I am still scratching my head about the UAS sending a 199. While
I see no reason to forbid it I'm trying to find a situation when it
would be helpful. The only case I have been able to think of is:

   There is a forking proxy upstream of the UAS, but it doesn't
   support 199. Yet it may delay the final response while it
   tries other forks. So the UAS sending the 199 makes up for
   the inadequacy of the forking proxy.

The trouble with this is that it is inefficient in many cases. If each
proxy that forks also supports the sending of 199, and the UAS doesn't
send one, then we avoid sending the message except in cases where it is
helpful. If the UAS always sends it we have increased the message count
by one for all failing calls. And the majority of failing calls get no
benefit from a 199.

This is one of those cases that could probably benefit from simulation
to judge the merits. My gut feel is that the costs of the UAS sending
the 199 outweigh the benefits. Barring some analysis to the contrary I
think I would say that a UAS SHOULD NOT send a 199. Or perhaps MAY send,
but without any guidance of when.

        Paul

Dale.Worley@comcast.net wrote:
> Here are some comments on this I-D.  I don't think that any of them
> modify our understanding of the proposal.  I guess this is very terse;
> please use these suggestions as you see fit.
>
> * Abstract
>
> Clarify:
>
>    This specification defines a new SIP response code, 199 Early
>    Dialog Terminated, which a SIP entity can use to indicate upstream
>    that an early dialog has been terminated.  The response code can be
>    used by a SIP entity (a SIP proxy or UAS), to inform the UAC that
>    an early dialog has been terminated.  As the 199 response is a
>    provisional response, often the UAC will receive it before
>    receiving a final response to its request.
>
> * Section 1
>
>    When a forking proxy receives a non-2xx final response, it does not
>    always immediately forward the response upstream towards the sender
>    of the associated request.  Instead, the forking proxy "stores" it
>    and waits for further final responses from remote destinations =
where
>    the forked request was forwarded.
>
> The text does not allow for serial forking, only parallel forking,
> which might be confusing to people who are more aware of serial
> forking.  So amend the last sentence to:
>
>    Instead, the forking proxy "stores" it and waits for further final
>    responses from remote destinations where the forked request was
>    forwarded, and possibly forwards the request to additional
>    destinations.
>
> Also, Avoid using "..." except when indicating that the word is used
> in a non-standard way or that one is discussing the word itself:
>
>    Instead, the forking proxy "stores" it [...]
>
> should be:
>
>    Instead, the forking proxy stores it [...]
>
> * Section 3.
>
>    REQ 1: It must be possible to indicate to the UAC that an early
>    dialog has been terminated before a final response is sent.
>
> Oddly, though everyone thinks they understand this sentence, it
> doesn't really say what we want.  To really ensure that the UAC
> understands the early dialog has been terminated before the UAS sends
> a final response, the UAS would have to send 199 *and get the PRACK
> saying that it was received* before sending a final response.  This =
may
> well be worse than the present state of affairs.
>
> What we really mean is:
>
>    REQ 1: It must be possible for a UAS or proxy to send an indication
>    to the UAC that an early dialog has been terminated, which
>    indication will not be delayed waiting for a final response from
>    any UAS.
>
> * Section 4
>
>    When a client receives a 199 Early Dialog Terminated provisional
>    response it MAY release resources and procedures associated with =
the
>    early dialog the 199 response is received on.
>
> Amend to clarify the meaning of 199 independently of the consequences
> of that meaning:
>
>    When a client receives a 199 Early Dialog Terminated provisional
>    response, it has positive knowledge that the UAS that created the
>    early dialog has terminated it, and so the client MAY release
>    resources and procedures associated with the early dialog for which
>    the 199 response is received.
>
> * Section 4
>
> Add this paragraph before the last paragraph:
>
>    A UAC that receives a 199 response for an early dialog MUST NOT
>    send any further requests on that dialog, except for a PRACK if the
>    199 response was sent reliably, and PRACKs for other reliable
>    provisional responses that are to be acknowledged.
>
> This is copied from section 7 (Backward Compatibility), but since it
> is a constraint on UACs, it should also be stated in this section.
>
> * Section 4
>
>    The UAC MAY use additional information (for example the final
>    response which triggered the 199 response) received in the 199
>    response when initiating new sessions, if it is possible to avoid =
the
>    new session initiation request to be rejected.
>
> What is the meaning of the final clause?
>
> * Section 5
>
> Add "early" as marked:
>
>    When a UAS wants to terminate *an early* dialog it sends a non-2xx
>    SIP final response, as specified in [RFC3261].  In addition, prior
>    to sending the non-2xx SIP final response, the UAS MAY send a 199
>    response to indicate that the dialog is being terminated.
>
> This clarifies that 199 cannot be used to terminate a confirmed
> dialog.
>
> (The reason a UAS may want to send a 199 is to provide this
> information even when the upstream proxy does not support 199.)
>
> There is a subtlty to the second sentence:
>
> Sending the 199 *before* the final response is not necessary due to
> the semantics of 199 (as it will probably reach the UAC first anyway,
> and it would not matter if it did not), but rather to prevent the
> upstream proxy from generating a 199 based on the final response, and
> then to transmit the UAS's 199 immediately after -- if the UAS sends
> the 199 first, any upstream proxy will (most likely) see the 199
> before seeing the final response (which might induce the proxy to send
> its own 199).
>
> * Section 6
>
>    When a forking proxy receives a non-2xx final response which
>    terminates one or more early dialogs and the proxy does not intend =
to
>    forward the final response immediately (due to the rules for a
>    forking proxy) it MUST send a 199 provisional response, for each
>    associated early dialog that it can associate with the final
>    response, upstream towards the sender of the associated request.
>
> Clarify, and note that if the proxy has already passed a 199 for an
> early dialog, it need not generate one itself.  Also, I've reduced
> MUST to SHOULD, because (as the following NOTE explains), the proxy
> may not know of all early dialogs which are terminated by the final
> response.
>
>    [...] it SHOULD initiate a 199 provisional response upstream
>    towards the sender of the associated request, for each terminated
>    early dialog, excepting those for which it has already transmitted
>    a 199 response.
>
> Note that the "already transmitted 199 responses" may have come from
> downstream proxies as well as downstream UASs, so this consideration
> applies even if we do not allow UASs to send 199s.
>
> * Section 7
>
>    Since all SIP entities involved in a session setup do not =
necessarily
>    support the specific meaning of the 199 Early Dialog Terminated
>    provisional response, the sender of the response MUST be prepared =
to
>    receive SIP requests associated with the dialog for which the 199
>    response was sent.
>
> This situation can also result due to race conditions.  Amend to:
>
>    Because of race conditions and because the client may not support =
the
>    specific meaning of the 199 provisional response, the sender of the
>    response MUST be prepared to receive SIP requests associated with
>    the dialog for which the 199 response was sent.
>
> And add a qualifier to this sentence:
>
>    If such request is received, and the receiver maintains state of
>    the dialogs, the receiver MUST reply to such requests with a 481
>    final response (unless another error response is more appropriate).
>
> * Section 7
>
> Remove "..." from this sentence and add "even" as marked:
>
>    The 199 Early Dialog Terminated response code MUST NOT replace a
>    final response.  A final response MUST always be sent, *even* after
>    one or many 199 Early Dialog Terminated provisional responses have
>    been sent.
>
> * Section 8
>
> Clarify this sentence to:
>
>    The 199 Early Dialog Terminated response code allows a SIP entity
>    to indicate to upstream entities that a specific early dialog has
>    been terminated, before a final response reaches the upstream
>    entities.
>
> Add the following:
>
>    The early dialog that has been terminated is identified by the =
Call-Id
>    header and the to-tag value of the 199 response.
>
>    Future standards may define optional additional information carried
>    in the headers and/or body of the response.
>
> * Section 9
>
>    A 199 Early Dialog Terminated provisional response MUST NOT contain =
a
>    new SDP offer/answer message body, but the sender of the response =
MAY
>    insert a copy of a previously sent offer/answer message body as
>    otherwise allowed by the offer/answer rules for a provisional
>    response.
>
> This rule seems excessively strict, as it seems OK to send a new SDP
> answer in a 199, even if the dialog will be immediately terminated.
> Certainly, the UAC cannot tell the difference, as there could have
> been a previous but lost 1xx response carrying the same SDP.
>
> The interaction of 199 and SDP probably needs further study...
>
> * Section 11
>
> Nit:  Use "reject" here rather than "rejects":
>
>    UAS_2 and UAS_3 *reject* the INVITE by sending a 4xx error =
response.
>
> Here is a modification of the example, showing how a UAS can send a
> 199, and how the proxy handles it.  (You may or may not want to use
> this.)  I've also added the ACK's for the 4xx's, which were missing in
> the original.
>
>    [...] UAS_2 and UAS_3 reject the INVITE by sending a 4xx error
>    response.  UAS_3 sends a 199 with its 4xx.  When P1 receives the
>    4xx response from UAS_2 it immediately sends a 199 Early Dialog
>    Terminated associated with UAS_2's dialog towards the UAC.  P1
>    transmits UAS_3's 199, and so does not initiate one itself for
>    UAS_3's dialog when it receives UAS_3's 4xx response.
>
>       UAC               P1                  UAS_2    UAS_3    UAS_4
>
>        --- INVITE ------>
>                          --- INVITE (leg 2) ->
>                          --- INVITE (leg 3) ---------->
>                          --- INVITE (leg 4) ------------------->
>                          <-- 18x (leg 2) -----
>        <-- 18x (leg 2) --
>                          <-- 18x (leg 3) --------------
>        <-- 18x (leg 3) --
>                          <-- 18x (leg 4) -----------------------
>        <-- 18x (leg 4) --
>                          <-- 4xx (leg 2) -----
>        <-- 199 (leg 2) --
>                          --- ACK (leg 2) ---->
>                          <-- 199 (leg 3) --------------
>        <-- 199 (leg 3) --
>                          <-- 4xx (leg 3) --------------
>                          --- ACK (leg 3) ------------->
>                          <-- 200 (leg 4) -----------------------
>        <-- 200 (leg 4) --
>        --- ACK (leg 4) ->
>                          --- ACK (leg 4) ---------------------->
>
> Dale
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip





.


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>RE: [Sip] Draft submission: =
draft-ietf-sip-199-00</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2900.3314" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D812122109-16062008><FONT =
face=3DVerdana=20
color=3D#0000ff size=3D2>Thanks Christer,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D812122109-16062008><FONT =
face=3DVerdana=20
color=3D#0000ff size=3D2>We have a query on our basic understanding of =
this=20
draft.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D812122109-16062008><FONT =
face=3DVerdana=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D812122109-16062008><FONT =
face=3DVerdana=20
color=3D#0000ff size=3D2>I too had the same query as Ashish, I can =
rephrase it for=20
all of us.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D812122109-16062008><FONT =
face=3DVerdana=20
color=3D#0000ff size=3D2>What we mean to ask is that, considering the =
scenario shown=20
by the SIP msg flow below, <STRONG>at what point </STRONG>does the proxy =
decide=20
to send 199 response <STRONG>and on which </STRONG>call=20
legs.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D812122109-16062008><FONT =
face=3DVerdana=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DVerdana><FONT =
color=3D#0000ff><FONT=20
size=3D2><SPAN class=3D812122109-16062008>For e.g. in the scenario =
below, when proxy=20
receives 403 on leg 2, and it is yet to receive responses on leg 3 and=20
4.</SPAN>&nbsp;<SPAN class=3D812122109-16062008>In this case, the proxy =
cannot=20
send a 199 to the UAC (is my understanding, please correct if=20
wrong).</SPAN></FONT></FONT></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DVerdana><FONT =
color=3D#0000ff><FONT=20
size=3D2><SPAN class=3D812122109-16062008>Also after having received 5xx =
and 6xx on=20
the leg 3 and 4, there is no use sending 199 now because the best final =
response=20
sent to UAC is sufficient to indicate that the leg 3 and 4 are=20
terminated.</SPAN></FONT></FONT></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DVerdana><FONT =
color=3D#0000ff><FONT=20
size=3D2><SPAN =
class=3D812122109-16062008></SPAN></FONT></FONT></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DVerdana><FONT =
color=3D#0000ff><FONT=20
size=3D2><SPAN =
class=3D812122109-16062008></SPAN></FONT></FONT></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DVerdana><FONT =
color=3D#0000ff><FONT=20
size=3D2><SPAN class=3D812122109-16062008>Please correct us if we are =
missing some=20
hidden lines in the draft w.r.t our=20
understanding.</SPAN></FONT></FONT></FONT></DIV>
<DIV><FONT face=3DVerdana color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV><SPAN class=3D812122109-16062008><FONT face=3DVerdana =
color=3D#0000ff=20
size=3D2>Thanks,</FONT></SPAN></DIV>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#0000ff size=3D2>Best=20
Regards,</FONT></DIV>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#0000ff =
size=3D2>Rekha</FONT></DIV>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#0000ff size=3D2>(ESN :=20
6-877-5834)</FONT></DIV>
<DIV align=3Dleft><FONT face=3DVerdana color=3D#0000ff size=3D2>(PSTN :=20
81168)</FONT></DIV>
<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> sip-bounces@ietf.org=20
[mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Christer=20
Holmberg<BR><B>Sent:</B> Monday, June 16, 2008 1:42 PM<BR><B>To:</B> =
Ashish=20
Saxena (WT01 - Telecom Equipment); pkyzivat@cisco.com;=20
Dale.Worley@comcast.net<BR><B>Cc:</B> sip@ietf.org<BR><B>Subject:</B> =
Re: [Sip]=20
Draft submission: draft-ietf-sip-199-00<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D473410708-16062008>Hi,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D473410708-16062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D473410708-16062008>The=20
way the "best" final response code is chosen is described in RFC 3261, =
and the=20
intention of 199 is not to change that.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D473410708-16062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D473410708-16062008>Regarding whether a 199 will be sent for leg =
4: if the=20
proxy is now going to send a "best" final response I guess it doesn't =
need to=20
send 199 for leg 4.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D473410708-16062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D473410708-16062008>Regards,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D473410708-16062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D473410708-16062008>Christer</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D473410708-16062008></SPAN></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>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
[mailto:sip-bounces@ietf.org] <B>On Behalf Of=20
</B>ashish.hsaxena@wipro.com<BR><B>Sent:</B> 16. kes=E4kuuta 2008=20
10:32<BR><B>To:</B> pkyzivat@cisco.com; =
Dale.Worley@comcast.net<BR><B>Cc:</B>=20
sip@ietf.org<BR><B>Subject:</B> Re: [Sip] Draft submission:=20
draft-ietf-sip-199-00<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV><SPAN class=3D937291707-16062008><FONT face=3DVerdana =
color=3D#800000=20
size=3D2>Referring to introduction part in the =
draft.</FONT></SPAN></DIV>
<DIV><SPAN class=3D937291707-16062008>&nbsp;&nbsp;&nbsp;" At some point =
the proxy=20
uses a&nbsp;specified mechanism to determine the "best" final response =
code,=20
and&nbsp;forwards that final response upstream towards the sender of=20
the&nbsp;associated request."</SPAN></DIV>
<DIV><SPAN class=3D937291707-16062008><FONT face=3DVerdana =
color=3D#800000=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D937291707-16062008><FONT face=3DVerdana =
color=3D#800000=20
size=3D2>Considering a scenario where all the forks have failed in =
yielding a 2xx=20
response from different UASes (UAS2 , 3 &amp; 4 in the diagram) with =
different=20
error codes, how will the proxy determine the "best" final response and =
to which=20
leg would it send the best final response. What would happen after 199 =
to leg=20
4?</FONT></SPAN></DIV>
<DIV><SPAN class=3D937291707-16062008><FONT face=3DVerdana =
color=3D#800000=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D937291707-16062008>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
UAC&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
P1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
UAS_2&nbsp;&nbsp;&nbsp; UAS_3&nbsp;&nbsp;&nbsp;=20
UAS_4<BR><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; --- INVITE=20
------&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;=20
--- INVITE (leg 2)=20
-&gt;<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;=20
--- INVITE (leg 3)=20
----------&gt;<BR>&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;=20
--- INVITE (leg 4)=20
-------------------&gt;<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;=20
&lt;-- 18x (leg 2) -----<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;-- =
18x (leg=20
2)=20
--<BR>&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;=20
&lt;-- 18x (leg 3) =
--------------<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;--=20
18x (leg 3)=20
--<BR>&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;=20
&lt;-- 18x (leg 4)=20
-----------------------<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;-- =
18x (leg=20
4)=20
--<BR>&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;=20
&lt;-- <STRONG>4xx</STRONG> (leg 2)=20
-----<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;-- 199 (leg 2)=20
--<BR>&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;=20
&lt;-- <STRONG>5xx </STRONG>(leg 3)=20
--------------<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;-- 199 (leg =
3)=20
--<BR>&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;=20
&lt;--&nbsp;<STRONG>6xx </STRONG>(leg 4)=20
-----------------------<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&lt;--&nbsp;199=20
(leg 4) --<BR></SPAN></DIV><!-- Converted from text/rtf format -->
<P><SPAN lang=3Den-us><FONT face=3DVerdana color=3D#800000 size=3D2>Best =

Regards,</FONT></SPAN> <BR><SPAN lang=3Den-us><FONT face=3DVerdana =
color=3D#800000=20
size=3D2>Ashish Saxena</FONT></SPAN> <BR><SPAN lang=3Den-us><FONT =
face=3DVerdana=20
color=3D#800000 size=3D2>877-5570</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> sip-bounces@ietf.org=20
[mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Attila =
Sipos<BR><B>Sent:</B>=20
Saturday, June 14, 2008 9:30 PM<BR><B>To:</B> Paul Kyzivat;=20
Dale.Worley@comcast.net<BR><B>Cc:</B> sip@ietf.org<BR><B>Subject:</B> =
Re: [Sip]=20
Draft submission: draft-ietf-sip-199-00<BR></FONT><BR></DIV>
<DIV></DIV><!-- Converted from text/plain format --><BR>
<P><FONT size=3D2>&gt;&gt;However I am still scratching my head about =
the UAS=20
sending a 199. While<BR>&gt;&gt;I see no reason to forbid it I'm trying =
to find=20
a situation when it<BR>&gt;&gt;would be helpful.<BR><BR>It may or may =
not be=20
helpful but as you say, you can<BR>see no reason to forbid =
it.<BR><BR>Why=20
restrict it if the restriction isn't needed?<BR><BR>As someone has =
pointed out,=20
a B2BUA scenario might benefit<BR>from 199.&nbsp; In fact, any kind of =
SIP=20
gateway might benefit.<BR><BR>&gt; This is one of those cases that could =

probably benefit from<BR>&gt; simulation to judge the merits. My gut =
feel is=20
that the costs<BR>&gt; of the UAS sending the 199 outweigh the=20
benefits.<BR><BR>If the UAS wants to send it, that is a decision =
for<BR>them -=20
not something that needs to be prevented.<BR><BR>SIP forking, in the =
technical=20
sense of sending a<BR>response with a different to-tag, is always =
legal.<BR>So,=20
any UA which has used forking<BR>should also be able to use =
199.<BR><BR>I don't=20
think we need to include any specific text about UAS<BR>except to say =
that a UAS=20
can use it if the UAS has created<BR>the=20
forking.<BR><BR>Regards,<BR><BR>Attila<BR><BR><BR><BR>-----Original=20
Message-----<BR>From: sip-bounces@ietf.org on behalf of Paul =
Kyzivat<BR>Sent:=20
Fri 13/06/2008 15:07<BR>To: Dale.Worley@comcast.net<BR>Cc:=20
sip@ietf.org<BR>Subject: Re: [Sip] Draft submission:=20
draft-ietf-sip-199-00<BR><BR>This all sounds like good =
stuff.<BR><BR>However I=20
am still scratching my head about the UAS sending a 199. While<BR>I see =
no=20
reason to forbid it I'm trying to find a situation when it<BR>would be =
helpful.=20
The only case I have been able to think of is:<BR><BR>&nbsp;&nbsp; There =
is a=20
forking proxy upstream of the UAS, but it doesn't<BR>&nbsp;&nbsp; =
support 199.=20
Yet it may delay the final response while it<BR>&nbsp;&nbsp; tries other =
forks.=20
So the UAS sending the 199 makes up for<BR>&nbsp;&nbsp; the inadequacy =
of the=20
forking proxy.<BR><BR>The trouble with this is that it is inefficient in =
many=20
cases. If each<BR>proxy that forks also supports the sending of 199, and =
the UAS=20
doesn't<BR>send one, then we avoid sending the message except in cases =
where it=20
is<BR>helpful. If the UAS always sends it we have increased the message=20
count<BR>by one for all failing calls. And the majority of failing calls =
get=20
no<BR>benefit from a 199.<BR><BR>This is one of those cases that could =
probably=20
benefit from simulation<BR>to judge the merits. My gut feel is that the =
costs of=20
the UAS sending<BR>the 199 outweigh the benefits. Barring some analysis =
to the=20
contrary I<BR>think I would say that a UAS SHOULD NOT send a 199. Or =
perhaps MAY=20
send,<BR>but without any guidance of=20
when.<BR><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
Paul<BR><BR>Dale.Worley@comcast.net wrote:<BR>&gt; Here are some =
comments on=20
this I-D.&nbsp; I don't think that any of them<BR>&gt; modify our =
understanding=20
of the proposal.&nbsp; I guess this is very terse;<BR>&gt; please use =
these=20
suggestions as you see fit.<BR>&gt;<BR>&gt; * Abstract<BR>&gt;<BR>&gt;=20
Clarify:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; This specification defines a =
new SIP=20
response code, 199 Early<BR>&gt;&nbsp;&nbsp;&nbsp; Dialog Terminated, =
which a=20
SIP entity can use to indicate upstream<BR>&gt;&nbsp;&nbsp;&nbsp; that =
an early=20
dialog has been terminated.&nbsp; The response code can=20
be<BR>&gt;&nbsp;&nbsp;&nbsp; used by a SIP entity (a SIP proxy or UAS), =
to=20
inform the UAC that<BR>&gt;&nbsp;&nbsp;&nbsp; an early dialog has been=20
terminated.&nbsp; As the 199 response is a<BR>&gt;&nbsp;&nbsp;&nbsp; =
provisional=20
response, often the UAC will receive it before<BR>&gt;&nbsp;&nbsp;&nbsp; =

receiving a final response to its request.<BR>&gt;<BR>&gt; * Section=20
1<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; When a forking proxy receives a =
non-2xx=20
final response, it does not<BR>&gt;&nbsp;&nbsp;&nbsp; always immediately =
forward=20
the response upstream towards the sender<BR>&gt;&nbsp;&nbsp;&nbsp; of =
the=20
associated request.&nbsp; Instead, the forking proxy "stores"=20
it<BR>&gt;&nbsp;&nbsp;&nbsp; and waits for further final responses from =
remote=20
destinations where<BR>&gt;&nbsp;&nbsp;&nbsp; the forked request was=20
forwarded.<BR>&gt;<BR>&gt; The text does not allow for serial forking, =
only=20
parallel forking,<BR>&gt; which might be confusing to people who are =
more aware=20
of serial<BR>&gt; forking.&nbsp; So amend the last sentence=20
to:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; Instead, the forking proxy =
"stores" it and=20
waits for further final<BR>&gt;&nbsp;&nbsp;&nbsp; responses from remote=20
destinations where the forked request was<BR>&gt;&nbsp;&nbsp;&nbsp; =
forwarded,=20
and possibly forwards the request to =
additional<BR>&gt;&nbsp;&nbsp;&nbsp;=20
destinations.<BR>&gt;<BR>&gt; Also, Avoid using "..." except when =
indicating=20
that the word is used<BR>&gt; in a non-standard way or that one is =
discussing=20
the word itself:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; Instead, the forking =
proxy=20
"stores" it [...]<BR>&gt;<BR>&gt; should =
be:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp;=20
Instead, the forking proxy stores it [...]<BR>&gt;<BR>&gt; * Section=20
3.<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; REQ 1: It must be possible to =
indicate to=20
the UAC that an early<BR>&gt;&nbsp;&nbsp;&nbsp; dialog has been =
terminated=20
before a final response is sent.<BR>&gt;<BR>&gt; Oddly, though everyone =
thinks=20
they understand this sentence, it<BR>&gt; doesn't really say what we =
want.&nbsp;=20
To really ensure that the UAC<BR>&gt; understands the early dialog has =
been=20
terminated before the UAS sends<BR>&gt; a final response, the UAS would =
have to=20
send 199 *and get the PRACK<BR>&gt; saying that it was received* before =
sending=20
a final response.&nbsp; This may<BR>&gt; well be worse than the present =
state of=20
affairs.<BR>&gt;<BR>&gt; What we really mean=20
is:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; REQ 1: It must be possible for a =
UAS or=20
proxy to send an indication<BR>&gt;&nbsp;&nbsp;&nbsp; to the UAC that an =
early=20
dialog has been terminated, which<BR>&gt;&nbsp;&nbsp;&nbsp; indication =
will not=20
be delayed waiting for a final response from<BR>&gt;&nbsp;&nbsp;&nbsp; =
any=20
UAS.<BR>&gt;<BR>&gt; * Section 4<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; When =
a client=20
receives a 199 Early Dialog Terminated =
provisional<BR>&gt;&nbsp;&nbsp;&nbsp;=20
response it MAY release resources and procedures associated with=20
the<BR>&gt;&nbsp;&nbsp;&nbsp; early dialog the 199 response is received=20
on.<BR>&gt;<BR>&gt; Amend to clarify the meaning of 199 independently of =
the=20
consequences<BR>&gt; of that meaning:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; =
When a=20
client receives a 199 Early Dialog Terminated=20
provisional<BR>&gt;&nbsp;&nbsp;&nbsp; response, it has positive =
knowledge that=20
the UAS that created the<BR>&gt;&nbsp;&nbsp;&nbsp; early dialog has =
terminated=20
it, and so the client MAY release<BR>&gt;&nbsp;&nbsp;&nbsp; resources =
and=20
procedures associated with the early dialog for =
which<BR>&gt;&nbsp;&nbsp;&nbsp;=20
the 199 response is received.<BR>&gt;<BR>&gt; * Section =
4<BR>&gt;<BR>&gt; Add=20
this paragraph before the last =
paragraph:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; A=20
UAC that receives a 199 response for an early dialog MUST=20
NOT<BR>&gt;&nbsp;&nbsp;&nbsp; send any further requests on that dialog, =
except=20
for a PRACK if the<BR>&gt;&nbsp;&nbsp;&nbsp; 199 response was sent =
reliably, and=20
PRACKs for other reliable<BR>&gt;&nbsp;&nbsp;&nbsp; provisional =
responses that=20
are to be acknowledged.<BR>&gt;<BR>&gt; This is copied from section 7 =
(Backward=20
Compatibility), but since it<BR>&gt; is a constraint on UACs, it should =
also be=20
stated in this section.<BR>&gt;<BR>&gt; * Section=20
4<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; The UAC MAY use additional =
information (for=20
example the final<BR>&gt;&nbsp;&nbsp;&nbsp; response which triggered the =
199=20
response) received in the 199<BR>&gt;&nbsp;&nbsp;&nbsp; response when =
initiating=20
new sessions, if it is possible to avoid the<BR>&gt;&nbsp;&nbsp;&nbsp; =
new=20
session initiation request to be rejected.<BR>&gt;<BR>&gt; What is the =
meaning=20
of the final clause?<BR>&gt;<BR>&gt; * Section 5<BR>&gt;<BR>&gt; Add =
"early" as=20
marked:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; When a UAS wants to terminate =
*an=20
early* dialog it sends a non-2xx<BR>&gt;&nbsp;&nbsp;&nbsp; SIP final =
response,=20
as specified in [RFC3261].&nbsp; In addition, =
prior<BR>&gt;&nbsp;&nbsp;&nbsp; to=20
sending the non-2xx SIP final response, the UAS MAY send a=20
199<BR>&gt;&nbsp;&nbsp;&nbsp; response to indicate that the dialog is =
being=20
terminated.<BR>&gt;<BR>&gt; This clarifies that 199 cannot be used to =
terminate=20
a confirmed<BR>&gt; dialog.<BR>&gt;<BR>&gt; (The reason a UAS may want =
to send a=20
199 is to provide this<BR>&gt; information even when the upstream proxy =
does not=20
support 199.)<BR>&gt;<BR>&gt; There is a subtlty to the second=20
sentence:<BR>&gt;<BR>&gt; Sending the 199 *before* the final response is =
not=20
necessary due to<BR>&gt; the semantics of 199 (as it will probably reach =
the UAC=20
first anyway,<BR>&gt; and it would not matter if it did not), but rather =
to=20
prevent the<BR>&gt; upstream proxy from generating a 199 based on the =
final=20
response, and<BR>&gt; then to transmit the UAS's 199 immediately after =
-- if the=20
UAS sends<BR>&gt; the 199 first, any upstream proxy will (most likely) =
see the=20
199<BR>&gt; before seeing the final response (which might induce the =
proxy to=20
send<BR>&gt; its own 199).<BR>&gt;<BR>&gt; * Section=20
6<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; When a forking proxy receives a =
non-2xx=20
final response which<BR>&gt;&nbsp;&nbsp;&nbsp; terminates one or more =
early=20
dialogs and the proxy does not intend to<BR>&gt;&nbsp;&nbsp;&nbsp; =
forward the=20
final response immediately (due to the rules for =
a<BR>&gt;&nbsp;&nbsp;&nbsp;=20
forking proxy) it MUST send a 199 provisional response, for=20
each<BR>&gt;&nbsp;&nbsp;&nbsp; associated early dialog that it can =
associate=20
with the final<BR>&gt;&nbsp;&nbsp;&nbsp; response, upstream towards the =
sender=20
of the associated request.<BR>&gt;<BR>&gt; Clarify, and note that if the =
proxy=20
has already passed a 199 for an<BR>&gt; early dialog, it need not =
generate one=20
itself.&nbsp; Also, I've reduced<BR>&gt; MUST to SHOULD, because (as the =

following NOTE explains), the proxy<BR>&gt; may not know of all early =
dialogs=20
which are terminated by the final<BR>&gt;=20
response.<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; [...] it SHOULD initiate a =
199=20
provisional response upstream<BR>&gt;&nbsp;&nbsp;&nbsp; towards the =
sender of=20
the associated request, for each terminated<BR>&gt;&nbsp;&nbsp;&nbsp; =
early=20
dialog, excepting those for which it has already=20
transmitted<BR>&gt;&nbsp;&nbsp;&nbsp; a 199 response.<BR>&gt;<BR>&gt; =
Note that=20
the "already transmitted 199 responses" may have come from<BR>&gt; =
downstream=20
proxies as well as downstream UASs, so this consideration<BR>&gt; =
applies even=20
if we do not allow UASs to send 199s.<BR>&gt;<BR>&gt; * Section=20
7<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; Since all SIP entities involved in a =
session=20
setup do not necessarily<BR>&gt;&nbsp;&nbsp;&nbsp; support the specific =
meaning=20
of the 199 Early Dialog Terminated<BR>&gt;&nbsp;&nbsp;&nbsp; provisional =

response, the sender of the response MUST be prepared=20
to<BR>&gt;&nbsp;&nbsp;&nbsp; receive SIP requests associated with the =
dialog for=20
which the 199<BR>&gt;&nbsp;&nbsp;&nbsp; response was =
sent.<BR>&gt;<BR>&gt; This=20
situation can also result due to race conditions.&nbsp; Amend=20
to:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; Because of race conditions and =
because the=20
client may not support the<BR>&gt;&nbsp;&nbsp;&nbsp; specific meaning of =
the 199=20
provisional response, the sender of the<BR>&gt;&nbsp;&nbsp;&nbsp; =
response MUST=20
be prepared to receive SIP requests associated =
with<BR>&gt;&nbsp;&nbsp;&nbsp;=20
the dialog for which the 199 response was sent.<BR>&gt;<BR>&gt; And add =
a=20
qualifier to this sentence:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; If such =
request is=20
received, and the receiver maintains state of<BR>&gt;&nbsp;&nbsp;&nbsp; =
the=20
dialogs, the receiver MUST reply to such requests with a=20
481<BR>&gt;&nbsp;&nbsp;&nbsp; final response (unless another error =
response is=20
more appropriate).<BR>&gt;<BR>&gt; * Section 7<BR>&gt;<BR>&gt; Remove =
"..." from=20
this sentence and add "even" as =
marked:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; The=20
199 Early Dialog Terminated response code MUST NOT replace=20
a<BR>&gt;&nbsp;&nbsp;&nbsp; final response.&nbsp; A final response MUST =
always=20
be sent, *even* after<BR>&gt;&nbsp;&nbsp;&nbsp; one or many 199 Early =
Dialog=20
Terminated provisional responses have<BR>&gt;&nbsp;&nbsp;&nbsp; been=20
sent.<BR>&gt;<BR>&gt; * Section 8<BR>&gt;<BR>&gt; Clarify this sentence=20
to:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; The 199 Early Dialog Terminated =
response=20
code allows a SIP entity<BR>&gt;&nbsp;&nbsp;&nbsp; to indicate to =
upstream=20
entities that a specific early dialog has<BR>&gt;&nbsp;&nbsp;&nbsp; been =

terminated, before a final response reaches the=20
upstream<BR>&gt;&nbsp;&nbsp;&nbsp; entities.<BR>&gt;<BR>&gt; Add the=20
following:<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; The early dialog that has =
been=20
terminated is identified by the Call-Id<BR>&gt;&nbsp;&nbsp;&nbsp; header =
and the=20
to-tag value of the 199 response.<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; =
Future=20
standards may define optional additional information=20
carried<BR>&gt;&nbsp;&nbsp;&nbsp; in the headers and/or body of the=20
response.<BR>&gt;<BR>&gt; * Section 9<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; =
A 199=20
Early Dialog Terminated provisional response MUST NOT contain=20
a<BR>&gt;&nbsp;&nbsp;&nbsp; new SDP offer/answer message body, but the =
sender of=20
the response MAY<BR>&gt;&nbsp;&nbsp;&nbsp; insert a copy of a previously =
sent=20
offer/answer message body as<BR>&gt;&nbsp;&nbsp;&nbsp; otherwise allowed =
by the=20
offer/answer rules for a provisional<BR>&gt;&nbsp;&nbsp;&nbsp;=20
response.<BR>&gt;<BR>&gt; This rule seems excessively strict, as it =
seems OK to=20
send a new SDP<BR>&gt; answer in a 199, even if the dialog will be =
immediately=20
terminated.<BR>&gt; Certainly, the UAC cannot tell the difference, as =
there=20
could have<BR>&gt; been a previous but lost 1xx response carrying the =
same=20
SDP.<BR>&gt;<BR>&gt; The interaction of 199 and SDP probably needs =
further=20
study...<BR>&gt;<BR>&gt; * Section 11<BR>&gt;<BR>&gt; Nit:&nbsp; Use =
"reject"=20
here rather than "rejects":<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; UAS_2 and =
UAS_3=20
*reject* the INVITE by sending a 4xx error response.<BR>&gt;<BR>&gt; =
Here is a=20
modification of the example, showing how a UAS can send a<BR>&gt; 199, =
and how=20
the proxy handles it.&nbsp; (You may or may not want to use<BR>&gt; =
this.)&nbsp;=20
I've also added the ACK's for the 4xx's, which were missing in<BR>&gt; =
the=20
original.<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp; [...] UAS_2 and UAS_3 reject =
the=20
INVITE by sending a 4xx error<BR>&gt;&nbsp;&nbsp;&nbsp; response.&nbsp; =
UAS_3=20
sends a 199 with its 4xx.&nbsp; When P1 receives =
the<BR>&gt;&nbsp;&nbsp;&nbsp;=20
4xx response from UAS_2 it immediately sends a 199 Early=20
Dialog<BR>&gt;&nbsp;&nbsp;&nbsp; Terminated associated with UAS_2's =
dialog=20
towards the UAC.&nbsp; P1<BR>&gt;&nbsp;&nbsp;&nbsp; transmits UAS_3's =
199, and=20
so does not initiate one itself for<BR>&gt;&nbsp;&nbsp;&nbsp; UAS_3's =
dialog=20
when it receives UAS_3's 4xx=20
response.<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
UAC&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
P1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
UAS_2&nbsp;&nbsp;&nbsp; UAS_3&nbsp;&nbsp;&nbsp;=20
UAS_4<BR>&gt;<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; --- =
INVITE=20
------&gt;<BR>&gt;&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;=20
--- INVITE (leg 2)=20
-&gt;<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;=20
--- INVITE (leg 3)=20
----------&gt;<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;=20
--- INVITE (leg 4)=20
-------------------&gt;<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&lt;-- 18x (leg 2) =
-----<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&lt;-- 18x (leg 2)=20
--<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
&lt;-- 18x (leg 3)=20
--------------<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;-- =
18x (leg=20
3)=20
--<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
&lt;-- 18x (leg 4)=20
-----------------------<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
 &lt;--=20
18x (leg 4)=20
--<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
&lt;-- 4xx (leg 2) =
-----<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&lt;-- 199 (leg 2)=20
--<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
--- ACK (leg 2)=20
----&gt;<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;=20
&lt;-- 199 (leg 3)=20
--------------<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;-- =
199 (leg=20
3)=20
--<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
&lt;-- 4xx (leg 3)=20
--------------<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;=20
--- ACK (leg 3)=20
-------------&gt;<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&lt;-- 200 (leg 4)=20
-----------------------<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
 &lt;--=20
200 (leg 4) --<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; --- ACK =
(leg 4)=20
-&gt;<BR>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;=20
--- ACK (leg 4) ----------------------&gt;<BR>&gt;<BR>&gt; Dale<BR>&gt;=20
_______________________________________________<BR>&gt; Sip mailing =
list&nbsp;=20
<A=20
href=3D"https://www.ietf.org/mailman/listinfo/sip">https://www.ietf.org/m=
ailman/listinfo/sip</A><BR>&gt;=20
This list is for 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>&gt;<BR>_______________________________________________<BR>Sip =
mailing=20
list&nbsp; <A=20
href=3D"https://www.ietf.org/mailman/listinfo/sip">https://www.ietf.org/m=
ailman/listinfo/sip</A><BR>This=20
list is for NEW development of the core SIP Protocol<BR>Use=20
sip-implementors@cs.columbia.edu for questions on current sip<BR>Use=20
sipping@ietf.org for new developments on the application of=20
sip<BR><BR></FONT></P><BR><BR>
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: <sreerekha.shenoy@wipro.com>, <ashish.hsaxena@wipro.com>,
	<pkyzivat@cisco.com>, <Dale.Worley@comcast.net>
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Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
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Hi, =


>We have a query on our basic understanding of this draft.
> =

>I too had the same query as Ashish, I can rephrase it for all of us.
>What we mean to ask is that, considering the scenario shown by the SIP msg=
 flow below, at what point does the proxy decide to send 199 response and o=
n which call legs.

A proxy would send a 199 as soon as it receives a final non-2xx response fo=
r a leg.

>For e.g. in the scenario below, when proxy receives 403 on leg 2, and it i=
s yet to receive responses on leg 3 and 4. In this case, the proxy cannot s=
end a 199 to the UAC (is my understanding, =

>please correct if wrong).

Yes, it can send 199 for leg 2. In SIP you can send as many provisional res=
ponses as you want.

>Also after having received 5xx and 6xx on the leg 3 and 4, there is no use=
 sending 199 now because the best final response sent to UAC is sufficient =
to indicate that the leg 3 and 4 are =

>terminated.

If the proxy is going to send a best final response immediately there is no=
 need to send 199. But, if the proxy is still waiting for final responses, =
it would send 199.

Regards,

Christer

 =


________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Chris=
ter Holmberg
Sent: Monday, June 16, 2008 1:42 PM
To: Ashish Saxena (WT01 - Telecom Equipment); pkyzivat@cisco.com; Dale.Worl=
ey@comcast.net
Cc: sip@ietf.org
Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00


 =

Hi,
 =

The way the "best" final response code is chosen is described in RFC 3261, =
and the intention of 199 is not to change that.
 =

Regarding whether a 199 will be sent for leg 4: if the proxy is now going t=
o send a "best" final response I guess it doesn't need to send 199 for leg =
4.
 =

Regards,
 =

Christer
 =

 =

 =


________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of ashis=
h.hsaxena@wipro.com
Sent: 16. kes=E4kuuta 2008 10:32
To: pkyzivat@cisco.com; Dale.Worley@comcast.net
Cc: sip@ietf.org
Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00


Referring to introduction part in the draft.
   " At some point the proxy uses a specified mechanism to determine the "b=
est" final response code, and forwards that final response upstream towards=
 the sender of the associated request."
 =

Considering a scenario where all the forks have failed in yielding a 2xx re=
sponse from different UASes (UAS2 , 3 & 4 in the diagram) with different er=
ror codes, how will the proxy determine the "best" final response and to wh=
ich leg would it send the best final response. What would happen after 199 =
to leg 4?
 =

      UAC               P1                  UAS_2    UAS_3    UAS_4

       --- INVITE ------>
                         --- INVITE (leg 2) ->
                         --- INVITE (leg 3) ---------->
                         --- INVITE (leg 4) ------------------->
                         <-- 18x (leg 2) -----
       <-- 18x (leg 2) --
                         <-- 18x (leg 3) --------------
       <-- 18x (leg 3) --
                         <-- 18x (leg 4) -----------------------
       <-- 18x (leg 4) --
                         <-- 4xx (leg 2) -----
       <-- 199 (leg 2) --
                         <-- 5xx (leg 3) --------------
       <-- 199 (leg 3) --
                         <-- 6xx (leg 4) -----------------------
       <-- 199 (leg 4) --


Best Regards, =

Ashish Saxena =

877-5570 =


 =


________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Attil=
a Sipos
Sent: Saturday, June 14, 2008 9:30 PM
To: Paul Kyzivat; Dale.Worley@comcast.net
Cc: sip@ietf.org
Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00




>>However I am still scratching my head about the UAS sending a 199. While
>>I see no reason to forbid it I'm trying to find a situation when it
>>would be helpful.

It may or may not be helpful but as you say, you can
see no reason to forbid it.

Why restrict it if the restriction isn't needed?

As someone has pointed out, a B2BUA scenario might benefit
from 199.  In fact, any kind of SIP gateway might benefit.

> This is one of those cases that could probably benefit from
> simulation to judge the merits. My gut feel is that the costs
> of the UAS sending the 199 outweigh the benefits.

If the UAS wants to send it, that is a decision for
them - not something that needs to be prevented.

SIP forking, in the technical sense of sending a
response with a different to-tag, is always legal.
So, any UA which has used forking
should also be able to use 199.

I don't think we need to include any specific text about UAS
except to say that a UAS can use it if the UAS has created
the forking.

Regards,

Attila



-----Original Message-----
From: sip-bounces@ietf.org on behalf of Paul Kyzivat
Sent: Fri 13/06/2008 15:07
To: Dale.Worley@comcast.net
Cc: sip@ietf.org
Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00

This all sounds like good stuff.

However I am still scratching my head about the UAS sending a 199. While
I see no reason to forbid it I'm trying to find a situation when it
would be helpful. The only case I have been able to think of is:

   There is a forking proxy upstream of the UAS, but it doesn't
   support 199. Yet it may delay the final response while it
   tries other forks. So the UAS sending the 199 makes up for
   the inadequacy of the forking proxy.

The trouble with this is that it is inefficient in many cases. If each
proxy that forks also supports the sending of 199, and the UAS doesn't
send one, then we avoid sending the message except in cases where it is
helpful. If the UAS always sends it we have increased the message count
by one for all failing calls. And the majority of failing calls get no
benefit from a 199.

This is one of those cases that could probably benefit from simulation
to judge the merits. My gut feel is that the costs of the UAS sending
the 199 outweigh the benefits. Barring some analysis to the contrary I
think I would say that a UAS SHOULD NOT send a 199. Or perhaps MAY send,
but without any guidance of when.

        Paul

Dale.Worley@comcast.net wrote:
> Here are some comments on this I-D.  I don't think that any of them
> modify our understanding of the proposal.  I guess this is very terse;
> please use these suggestions as you see fit.
>
> * Abstract
>
> Clarify:
>
>    This specification defines a new SIP response code, 199 Early
>    Dialog Terminated, which a SIP entity can use to indicate upstream
>    that an early dialog has been terminated.  The response code can be
>    used by a SIP entity (a SIP proxy or UAS), to inform the UAC that
>    an early dialog has been terminated.  As the 199 response is a
>    provisional response, often the UAC will receive it before
>    receiving a final response to its request.
>
> * Section 1
>
>    When a forking proxy receives a non-2xx final response, it does not
>    always immediately forward the response upstream towards the sender
>    of the associated request.  Instead, the forking proxy "stores" it
>    and waits for further final responses from remote destinations where
>    the forked request was forwarded.
>
> The text does not allow for serial forking, only parallel forking,
> which might be confusing to people who are more aware of serial
> forking.  So amend the last sentence to:
>
>    Instead, the forking proxy "stores" it and waits for further final
>    responses from remote destinations where the forked request was
>    forwarded, and possibly forwards the request to additional
>    destinations.
>
> Also, Avoid using "..." except when indicating that the word is used
> in a non-standard way or that one is discussing the word itself:
>
>    Instead, the forking proxy "stores" it [...]
>
> should be:
>
>    Instead, the forking proxy stores it [...]
>
> * Section 3.
>
>    REQ 1: It must be possible to indicate to the UAC that an early
>    dialog has been terminated before a final response is sent.
>
> Oddly, though everyone thinks they understand this sentence, it
> doesn't really say what we want.  To really ensure that the UAC
> understands the early dialog has been terminated before the UAS sends
> a final response, the UAS would have to send 199 *and get the PRACK
> saying that it was received* before sending a final response.  This may
> well be worse than the present state of affairs.
>
> What we really mean is:
>
>    REQ 1: It must be possible for a UAS or proxy to send an indication
>    to the UAC that an early dialog has been terminated, which
>    indication will not be delayed waiting for a final response from
>    any UAS.
>
> * Section 4
>
>    When a client receives a 199 Early Dialog Terminated provisional
>    response it MAY release resources and procedures associated with the
>    early dialog the 199 response is received on.
>
> Amend to clarify the meaning of 199 independently of the consequences
> of that meaning:
>
>    When a client receives a 199 Early Dialog Terminated provisional
>    response, it has positive knowledge that the UAS that created the
>    early dialog has terminated it, and so the client MAY release
>    resources and procedures associated with the early dialog for which
>    the 199 response is received.
>
> * Section 4
>
> Add this paragraph before the last paragraph:
>
>    A UAC that receives a 199 response for an early dialog MUST NOT
>    send any further requests on that dialog, except for a PRACK if the
>    199 response was sent reliably, and PRACKs for other reliable
>    provisional responses that are to be acknowledged.
>
> This is copied from section 7 (Backward Compatibility), but since it
> is a constraint on UACs, it should also be stated in this section.
>
> * Section 4
>
>    The UAC MAY use additional information (for example the final
>    response which triggered the 199 response) received in the 199
>    response when initiating new sessions, if it is possible to avoid the
>    new session initiation request to be rejected.
>
> What is the meaning of the final clause?
>
> * Section 5
>
> Add "early" as marked:
>
>    When a UAS wants to terminate *an early* dialog it sends a non-2xx
>    SIP final response, as specified in [RFC3261].  In addition, prior
>    to sending the non-2xx SIP final response, the UAS MAY send a 199
>    response to indicate that the dialog is being terminated.
>
> This clarifies that 199 cannot be used to terminate a confirmed
> dialog.
>
> (The reason a UAS may want to send a 199 is to provide this
> information even when the upstream proxy does not support 199.)
>
> There is a subtlty to the second sentence:
>
> Sending the 199 *before* the final response is not necessary due to
> the semantics of 199 (as it will probably reach the UAC first anyway,
> and it would not matter if it did not), but rather to prevent the
> upstream proxy from generating a 199 based on the final response, and
> then to transmit the UAS's 199 immediately after -- if the UAS sends
> the 199 first, any upstream proxy will (most likely) see the 199
> before seeing the final response (which might induce the proxy to send
> its own 199).
>
> * Section 6
>
>    When a forking proxy receives a non-2xx final response which
>    terminates one or more early dialogs and the proxy does not intend to
>    forward the final response immediately (due to the rules for a
>    forking proxy) it MUST send a 199 provisional response, for each
>    associated early dialog that it can associate with the final
>    response, upstream towards the sender of the associated request.
>
> Clarify, and note that if the proxy has already passed a 199 for an
> early dialog, it need not generate one itself.  Also, I've reduced
> MUST to SHOULD, because (as the following NOTE explains), the proxy
> may not know of all early dialogs which are terminated by the final
> response.
>
>    [...] it SHOULD initiate a 199 provisional response upstream
>    towards the sender of the associated request, for each terminated
>    early dialog, excepting those for which it has already transmitted
>    a 199 response.
>
> Note that the "already transmitted 199 responses" may have come from
> downstream proxies as well as downstream UASs, so this consideration
> applies even if we do not allow UASs to send 199s.
>
> * Section 7
>
>    Since all SIP entities involved in a session setup do not necessarily
>    support the specific meaning of the 199 Early Dialog Terminated
>    provisional response, the sender of the response MUST be prepared to
>    receive SIP requests associated with the dialog for which the 199
>    response was sent.
>
> This situation can also result due to race conditions.  Amend to:
>
>    Because of race conditions and because the client may not support the
>    specific meaning of the 199 provisional response, the sender of the
>    response MUST be prepared to receive SIP requests associated with
>    the dialog for which the 199 response was sent.
>
> And add a qualifier to this sentence:
>
>    If such request is received, and the receiver maintains state of
>    the dialogs, the receiver MUST reply to such requests with a 481
>    final response (unless another error response is more appropriate).
>
> * Section 7
>
> Remove "..." from this sentence and add "even" as marked:
>
>    The 199 Early Dialog Terminated response code MUST NOT replace a
>    final response.  A final response MUST always be sent, *even* after
>    one or many 199 Early Dialog Terminated provisional responses have
>    been sent.
>
> * Section 8
>
> Clarify this sentence to:
>
>    The 199 Early Dialog Terminated response code allows a SIP entity
>    to indicate to upstream entities that a specific early dialog has
>    been terminated, before a final response reaches the upstream
>    entities.
>
> Add the following:
>
>    The early dialog that has been terminated is identified by the Call-Id
>    header and the to-tag value of the 199 response.
>
>    Future standards may define optional additional information carried
>    in the headers and/or body of the response.
>
> * Section 9
>
>    A 199 Early Dialog Terminated provisional response MUST NOT contain a
>    new SDP offer/answer message body, but the sender of the response MAY
>    insert a copy of a previously sent offer/answer message body as
>    otherwise allowed by the offer/answer rules for a provisional
>    response.
>
> This rule seems excessively strict, as it seems OK to send a new SDP
> answer in a 199, even if the dialog will be immediately terminated.
> Certainly, the UAC cannot tell the difference, as there could have
> been a previous but lost 1xx response carrying the same SDP.
>
> The interaction of 199 and SDP probably needs further study...
>
> * Section 11
>
> Nit:  Use "reject" here rather than "rejects":
>
>    UAS_2 and UAS_3 *reject* the INVITE by sending a 4xx error response.
>
> Here is a modification of the example, showing how a UAS can send a
> 199, and how the proxy handles it.  (You may or may not want to use
> this.)  I've also added the ACK's for the 4xx's, which were missing in
> the original.
>
>    [...] UAS_2 and UAS_3 reject the INVITE by sending a 4xx error
>    response.  UAS_3 sends a 199 with its 4xx.  When P1 receives the
>    4xx response from UAS_2 it immediately sends a 199 Early Dialog
>    Terminated associated with UAS_2's dialog towards the UAC.  P1
>    transmits UAS_3's 199, and so does not initiate one itself for
>    UAS_3's dialog when it receives UAS_3's 4xx response.
>
>       UAC               P1                  UAS_2    UAS_3    UAS_4
>
>        --- INVITE ------>
>                          --- INVITE (leg 2) ->
>                          --- INVITE (leg 3) ---------->
>                          --- INVITE (leg 4) ------------------->
>                          <-- 18x (leg 2) -----
>        <-- 18x (leg 2) --
>                          <-- 18x (leg 3) --------------
>        <-- 18x (leg 3) --
>                          <-- 18x (leg 4) -----------------------
>        <-- 18x (leg 4) --
>                          <-- 4xx (leg 2) -----
>        <-- 199 (leg 2) --
>                          --- ACK (leg 2) ---->
>                          <-- 199 (leg 3) --------------
>        <-- 199 (leg 3) --
>                          <-- 4xx (leg 3) --------------
>                          --- ACK (leg 3) ------------->
>                          <-- 200 (leg 4) -----------------------
>        <-- 200 (leg 4) --
>        --- ACK (leg 4) ->
>                          --- ACK (leg 4) ---------------------->
>
> Dale
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip





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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Rockson Li (zhengyli)" <zhengyli@cisco.com>,
	<sip@ietf.org>
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Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt
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Hi Rockson,
=20
Ok, now I understand. You are talking about the case where the UAC =
doesn't receive the 18x, and then gets confused when it receives the =
199. That can happen no matter whether it's a proxy or UAS that sends =
the 199.
=20
I guess that is something which could happen, yes, and a UAC could think =
a new early dialog has been created. As you say, a way to avoid it would =
be to send at least one of the 18x reliable before a 199 is sent.=20
=20
But, I am not sure we want to mandate that, because I don't think =
anything will break if this happens.
=20
Regards,
=20
Chrsiter
=20
=20

________________________________

From: Rockson Li (zhengyli) [mailto:zhengyli@cisco.com]=20
Sent: 16. kes=E4kuuta 2008 4:51
To: Christer Holmberg; sip@ietf.org
Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt


Hi Christer,
=20
It would be easier for UAS to tell if it sends non-100 1xx response =
creating early-dialog before sending 199.
However, it's a little bit difficult from UAC's perspective, since the =
non-100 1xx response might not be reliable.
=20
If UAC does not get , say 180, but get 199 first, it looks like it does =
not make any sense.
=20
One option is to mandate usage of 100rel, which might be too strict for =
this draft.
=20
Regards,
-Rockson
-------------------------------------------------------------------------=
------------------
Date: Sun, 15 Jun 2008 13:52:24 +0200
From: "Christer Holmberg" <christer.holmberg@ericsson.com>
Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt
To: "Rockson Zhengyu Li" <lzyiii@laszlomail.com>, <sip@ietf.org>
Message-ID:
 <CA9998CD4A020D418654FCDEF4E707DF06B19A2E@esealmw113.eemea.ericsson.se>
=20
Content-Type: text/plain; charset=3D"iso-8859-1"
=20
Hi,
=20
You wouldn't send a 199 unless you have already sent a non-100 1xx =
response, so the early dialog has already been created.
=20
Regards,
=20
Christer
=20
________________________________
=20
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =
Rockson Zhengyu Li
Sent: 15. kes?kuuta 2008 14:43
To: sip@ietf.org
Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt
=20

Hi,
=20

one questions on this draft.
=20

Since RFC3261 says, non-100 1xx response for INVITE cause early-dialog =
creation.
does this draft will normativly change the criteria, excluding 199?
=20

If, the answer is Yes, I think it need be stated explicitly in the =
draft.
=20
=20
=20
thanks
=20

Regards,
-Rockson =20
=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=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2800.1609" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D255073611-16062008>Hi=20
Rockson,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D255073611-16062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D255073611-16062008>Ok,=20
now I understand. You are talking about the case where the UAC doesn't =
receive=20
the 18x, and then gets confused when it receives the 199. That can =
happen no=20
matter whether it's a proxy or UAS that sends the =
199.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D255073611-16062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D255073611-16062008>I=20
guess that is something which could happen, yes, and a UAC could think a =
new=20
early dialog has been created.&nbsp;As you say,&nbsp;a way to =
avoid&nbsp;it=20
would be to send at least one of the 18x reliable before a 199 is sent.=20
</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D255073611-16062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D255073611-16062008>But, I=20
am not sure we want to mandate that, because I don't think anything will =
break=20
if this happens.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D255073611-16062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D255073611-16062008>Regards,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D255073611-16062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D255073611-16062008>Chrsiter</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><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> Rockson Li (zhengyli)=20
[mailto:zhengyli@cisco.com] <BR><B>Sent:</B> 16. kes=E4kuuta 2008=20
4:51<BR><B>To:</B> Christer Holmberg; sip@ietf.org<BR><B>Subject:</B> =
Re: [Sip]=20
I-D ACTION:draft-ietf-sip-199-00.txt<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D390394401-16062008>Hi=20
</SPAN></FONT><FONT face=3DArial size=3D2><SPAN=20
class=3D390394401-16062008>Christer,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D390394401-16062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D390394401-16062008>It =
would be easier=20
for UAS to tell if it sends non-100 1xx response creating early-dialog =
before=20
sending 199.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN =
class=3D390394401-16062008>However,&nbsp;it's=20
a&nbsp;little bit difficult from</SPAN></FONT><FONT face=3DArial =
size=3D2><SPAN=20
class=3D390394401-16062008>&nbsp;UAC's perspective, since the non-100 =
1xx response=20
might not be reliable.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D390394401-16062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D390394401-16062008>If UAC =
does not get=20
, say 180, but get 199 first, it looks like it does not make any=20
sense.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D390394401-16062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D390394401-16062008>One =
option is to=20
mandate usage of 100rel, which might be too strict for this=20
draft.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D390394401-16062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D390394401-16062008>Regards,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D390394401-16062008>-Rockson</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D390394401-16062008>----------------------------------------------=
---------------------------------------------</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Date: Sun, 15 Jun 2008 13:52:24 =
+0200<BR>From:=20
"Christer Holmberg" &lt;<A=20
href=3D"mailto:christer.holmberg@ericsson.com">christer.holmberg@ericsson=
.com</A>&gt;<BR>Subject:=20
Re: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt<BR>To: "Rockson Zhengyu =
Li"=20
&lt;<A=20
href=3D"mailto:lzyiii@laszlomail.com">lzyiii@laszlomail.com</A>&gt;,&nbsp=
;&lt;<A=20
href=3D"mailto:sip@ietf.org">sip@ietf.org</A>&gt;<BR>Message-ID:<BR>&nbsp=
;&lt;<A=20
href=3D"mailto:CA9998CD4A020D418654FCDEF4E707DF06B19A2E@esealmw113.eemea.=
ericsson.se">CA9998CD4A020D418654FCDEF4E707DF06B19A2E@esealmw113.eemea.er=
icsson.se</A>&gt;<BR>&nbsp;<BR>Content-Type:=20
text/plain; charset=3D"iso-8859-1"</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Hi,<BR>&nbsp;<BR>You wouldn't send a =
199 unless you=20
have already sent a non-100 1xx response, so the early dialog has =
already been=20
created.<BR>&nbsp;<BR>Regards,<BR>&nbsp;<BR>Christer</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial =
size=3D2>________________________________</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>From: <A=20
href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</A>=20
[mailto:sip-bounces@ietf.org] On Behalf Of Rockson Zhengyu Li<BR>Sent: =
15.=20
kes?kuuta 2008 14:43<BR>To: <A=20
href=3D"mailto:sip@ietf.org">sip@ietf.org</A><BR>Subject: Re: [Sip] I-D=20
ACTION:draft-ietf-sip-199-00.txt</FONT></DIV>
<DIV>&nbsp;</DIV><FONT face=3DArial size=3D2>
<DIV><BR>Hi,</DIV>
<DIV>&nbsp;</DIV>
<DIV><BR>one questions on this draft.</DIV>
<DIV>&nbsp;</DIV>
<DIV><BR>Since RFC3261 says, non-100 1xx response for INVITE cause =
early-dialog=20
creation.<BR>does this draft will normativly change the criteria, =
excluding=20
199?</DIV>
<DIV>&nbsp;</DIV>
<DIV><BR>If, the answer is Yes, I think it need be stated explicitly in =
the=20
draft.</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>thanks</DIV>
<DIV>&nbsp;</DIV>
<DIV><BR>Regards,<BR>-Rockson&nbsp; </DIV>
<DIV>&nbsp;</DIV>
<DIV></FONT>&nbsp;</DIV></BODY></HTML>

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_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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Thread-Topic: [Sip] draft-dotson-sip-mutual-auth-03
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>,
	<sip@ietf.org>
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Subject: Re: [Sip] draft-dotson-sip-mutual-auth-03
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Hi,

My position of the draft is supportive.

Regards,

Christer
 =


-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of DRAGE=
, Keith (Keith)
Sent: 12. kes=E4kuuta 2008 16:50
To: sip@ietf.org
Subject: Re: [Sip] draft-dotson-sip-mutual-auth-03

(As SIP WG cochair)

The document has now been updated based on comments received.

http://www.ietf.org/internet-drafts/draft-dotson-sip-mutual-auth-03.txt

I will give a few more days for responses to the question of support / no s=
upport.

Please read the draft and indicate your postion.

Regards

Keith =


> -----Original Message-----
> From: Stuart Hoggan [mailto:s.hoggan@CableLabs.com]
> Sent: Tuesday, June 10, 2008 11:01 PM
> To: DRAGE, Keith (Keith); sip@ietf.org
> Subject: RE: [Sip] draft-dotson-sip-mutual-auth-02
> =

> All,
> =

> We have updated the document to incorporate feedback received so far.
> It now contains a section that helps clarify the motivation.  =

> Let us know if you have further comments. =

> =

> -Stuart
> =

> > -----Original Message-----
> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
> Behalf Of
> > DRAGE, Keith (Keith)
> > Sent: Wednesday, June 04, 2008 3:48 AM
> > To: sip@ietf.org
> > Subject: Re: [Sip] draft-dotson-sip-mutual-auth-02
> > =

> > (As SIP WG cochair)
> > =

> > This has been raised in the SIP group a couple of times, and we have
> not
> > yet gained an idea of whether to proceed with it in the SIP
> WG or not.
> > =

> > There have been a couple of technical comments raised in
> the past from
> > the security experts; my understanding is that these have now been =

> > clarified.
> > =

> > I believe in the past there has been an interested
> community that says
> > this is useful. It has also been clarified that 3GPP would
> like to use
> > it in their specifications.
> > =

> > So could I ask the WG to look at this document, and formally to
> indicate
> > whether they believe some work with this scope should be
> progressed as
> a
> > SIP WG item.
> > =

> >
> http://www.ietf.org/internet-drafts/draft-dotson-sip-mutual-au
> th-02.txt
> > =

> > So if you support it please indicate to the SIP WG chairs.
> > =

> > If you have technical concerns (i.e. it should not proceed
> because it
> is
> > technically flawed beyond fixing) then please ideally mail
> the list,
> > although the SIP WG chairs will also accept input.
> > =

> > Responses please in 1 calendar week from this date.
> > =

> > Regards
> > =

> > Keith
> > =

> > =

> > ________________________________
> > =

> > 	From: sip-bounces@ietf.org
> [mailto:sip-bounces@ietf.org] On Behalf Of
> > Stuart Hoggan
> > 	Sent: Friday, May 02, 2008 5:15 PM
> > 	To: sip@ietf.org
> > 	Subject: [Sip] draft-dotson-sip-mutual-auth-02
> > =

> > =

> > =

> > 	Folks,
> > =

> > =

> > =

> > 	Based on the feedback received in Philadelphia, we have updated =

> > draft-dotson-sip-mutual-auth. The changes clarify the usage of the =

> > Proxy-Authentication-Info header, including support for multiple
> values.
> > =

> > =

> > =

> > =

> > 	Any further comments or suggestions will be
> appreciated. It would
> > also be nice to obtain WG feedback on pursuing this I-D as a WG =

> > document.
> > =

> > =

> > =

> > 	Regards,
> > =

> > 	Stuart
> > =

> > =

> > =

> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use =

> > sip-implementors@cs.columbia.edu for questions on current sip Use =

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

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


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From: Paul Kyzivat <pkyzivat@cisco.com>
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	<CA9998CD4A020D418654FCDEF4E707DF06B4D8AD@esealmw113.eemea.ericsson.se>
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Cc: sip@ietf.org, "Rockson Li \(zhengyli\)" <zhengyli@cisco.com>
Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt
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I don't understand the problem here.
If the UAC receives the 199 as the first provisional response in a new =

dialog, then it will in effect create a new dialog and then immediately =

terminate it based on the 199.

If the UAC doesn't understand the new semantics of 199 then it will =

treat it as an arbitrary 1xx response, and create a new dialog that then =

will have to time out. But that is no different than if the 18x response =

had not been lost.

	Paul

Christer Holmberg wrote:
> Hi Rockson,
>  =

> Ok, now I understand. You are talking about the case where the UAC =

> doesn't receive the 18x, and then gets confused when it receives the =

> 199. That can happen no matter whether it's a proxy or UAS that sends =

> the 199.
>  =

> I guess that is something which could happen, yes, and a UAC could think =

> a new early dialog has been created. As you say, a way to avoid it would =

> be to send at least one of the 18x reliable before a 199 is sent.
>  =

> But, I am not sure we want to mandate that, because I don't think =

> anything will break if this happens.
>  =

> Regards,
>  =

> Chrsiter
>  =

>  =

> =

> ------------------------------------------------------------------------
> *From:* Rockson Li (zhengyli) [mailto:zhengyli@cisco.com]
> *Sent:* 16. kes=E4kuuta 2008 4:51
> *To:* Christer Holmberg; sip@ietf.org
> *Subject:* Re: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt
> =

> Hi Christer,
>  =

> It would be easier for UAS to tell if it sends non-100 1xx response =

> creating early-dialog before sending 199.
> However, it's a little bit difficult from UAC's perspective, since the =

> non-100 1xx response might not be reliable.
>  =

> If UAC does not get , say 180, but get 199 first, it looks like it does =

> not make any sense.
>  =

> One option is to mandate usage of 100rel, which might be too strict for =

> this draft.
>  =

> Regards,
> -Rockson
> -------------------------------------------------------------------------=
------------------
> Date: Sun, 15 Jun 2008 13:52:24 +0200
> From: "Christer Holmberg" <christer.holmberg@ericsson.com =

> <mailto:christer.holmberg@ericsson.com>>
> Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt
> To: "Rockson Zhengyu Li" <lzyiii@laszlomail.com =

> <mailto:lzyiii@laszlomail.com>>, <sip@ietf.org <mailto:sip@ietf.org>>
> Message-ID:
>  <CA9998CD4A020D418654FCDEF4E707DF06B19A2E@esealmw113.eemea.ericsson.se =

> <mailto:CA9998CD4A020D418654FCDEF4E707DF06B19A2E@esealmw113.eemea.ericsso=
n.se>>
>  =

> Content-Type: text/plain; charset=3D"iso-8859-1"
>  =

> Hi,
>  =

> You wouldn't send a 199 unless you have already sent a non-100 1xx =

> response, so the early dialog has already been created.
>  =

> Regards,
>  =

> Christer
>  =

> ________________________________
>  =

> From: sip-bounces@ietf.org <mailto:sip-bounces@ietf.org> =

> [mailto:sip-bounces@ietf.org] On Behalf Of Rockson Zhengyu Li
> Sent: 15. kes?kuuta 2008 14:43
> To: sip@ietf.org <mailto:sip@ietf.org>
> Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt
>  =

> =

> Hi,
>  =

> =

> one questions on this draft.
>  =

> =

> Since RFC3261 says, non-100 1xx response for INVITE cause early-dialog =

> creation.
> does this draft will normativly change the criteria, excluding 199?
>  =

> =

> If, the answer is Yes, I think it need be stated explicitly in the draft.
>  =

>  =

>  =

> thanks
>  =

> =

> Regards,
> -Rockson =

>  =

>  =

> =

> =

> ------------------------------------------------------------------------
> =

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


From sip-bounces@ietf.org  Mon Jun 16 05:21:47 2008
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Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt
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Ya, I agree with Paul. I was about to write the similar thing that 199 will=
 be treated as per non 100 1xx response as 3261 says (in case UAC does not =
understand 199 message).  =



Best Regards,
Ashish Saxena


-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Paul =
Kyzivat
Sent: Monday, June 16, 2008 5:48 PM
To: Christer Holmberg
Cc: sip@ietf.org; Rockson Li (zhengyli)
Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt

I don't understand the problem here.
If the UAC receives the 199 as the first provisional response in a new dial=
og, then it will in effect create a new dialog and then immediately termina=
te it based on the 199.

If the UAC doesn't understand the new semantics of 199 then it will treat i=
t as an arbitrary 1xx response, and create a new dialog that then will have=
 to time out. But that is no different than if the 18x response had not bee=
n lost.

	Paul

Christer Holmberg wrote:
> Hi Rockson,
>  =

> Ok, now I understand. You are talking about the case where the UAC =

> doesn't receive the 18x, and then gets confused when it receives the =

> 199. That can happen no matter whether it's a proxy or UAS that sends =

> the 199.
>  =

> I guess that is something which could happen, yes, and a UAC could =

> think a new early dialog has been created. As you say, a way to avoid =

> it would be to send at least one of the 18x reliable before a 199 is sent.
>  =

> But, I am not sure we want to mandate that, because I don't think =

> anything will break if this happens.
>  =

> Regards,
>  =

> Chrsiter
>  =

>  =

> =

> ----------------------------------------------------------------------
> --
> *From:* Rockson Li (zhengyli) [mailto:zhengyli@cisco.com]
> *Sent:* 16. kes=E4kuuta 2008 4:51
> *To:* Christer Holmberg; sip@ietf.org
> *Subject:* Re: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt
> =

> Hi Christer,
>  =

> It would be easier for UAS to tell if it sends non-100 1xx response =

> creating early-dialog before sending 199.
> However, it's a little bit difficult from UAC's perspective, since the =

> non-100 1xx response might not be reliable.
>  =

> If UAC does not get , say 180, but get 199 first, it looks like it =

> does not make any sense.
>  =

> One option is to mandate usage of 100rel, which might be too strict =

> for this draft.
>  =

> Regards,
> -Rockson
> ----------------------------------------------------------------------
> ---------------------
> Date: Sun, 15 Jun 2008 13:52:24 +0200
> From: "Christer Holmberg" <christer.holmberg@ericsson.com =

> <mailto:christer.holmberg@ericsson.com>>
> Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt
> To: "Rockson Zhengyu Li" <lzyiii@laszlomail.com =

> <mailto:lzyiii@laszlomail.com>>, <sip@ietf.org <mailto:sip@ietf.org>>
> Message-ID:
>  =

> <CA9998CD4A020D418654FCDEF4E707DF06B19A2E@esealmw113.eemea.ericsson.se
> <mailto:CA9998CD4A020D418654FCDEF4E707DF06B19A2E@esealmw113.eemea.eric
> sson.se>>
>  =

> Content-Type: text/plain; charset=3D"iso-8859-1"
>  =

> Hi,
>  =

> You wouldn't send a 199 unless you have already sent a non-100 1xx =

> response, so the early dialog has already been created.
>  =

> Regards,
>  =

> Christer
>  =

> ________________________________
>  =

> From: sip-bounces@ietf.org <mailto:sip-bounces@ietf.org> =

> [mailto:sip-bounces@ietf.org] On Behalf Of Rockson Zhengyu Li
> Sent: 15. kes?kuuta 2008 14:43
> To: sip@ietf.org <mailto:sip@ietf.org>
> Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt
>  =

> =

> Hi,
>  =

> =

> one questions on this draft.
>  =

> =

> Since RFC3261 says, non-100 1xx response for INVITE cause early-dialog =

> creation.
> does this draft will normativly change the criteria, excluding 199?
>  =

> =

> If, the answer is Yes, I think it need be stated explicitly in the draft.
>  =

>  =

>  =

> thanks
>  =

> =

> Regards,
> -Rockson
>  =

>  =

> =

> =

> ----------------------------------------------------------------------
> --
> =

> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use =

> sip-implementors@cs.columbia.edu for questions on current sip Use =

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


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From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
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As you can never tell whether a UA is a UA by itself, or part of a
B2BUA, this must remain general UA behaviour.

The most you can do is add some informative material that essentially
says this is useful for B2BUAs only.

Regards

Keith 

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of Juha Heinanen
> Sent: Saturday, June 14, 2008 2:21 PM
> To: Christer Holmberg
> Cc: sip@ietf.org; Paul Kyzivat; Francois Audet
> Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
> 
> Christer Holmberg writes:
> 
>  >  Back-to-Back User Agent: A back-to-back user agent (B2BUA) is a
>  >          logical entity that receives a request and 
> processes it as a
>  >          user agent server (UAS).  In order to determine 
> how the request
>  >          should be answered, it acts as a user agent 
> client (UAC) and
>  >          generates requests.  Unlike a proxy server, it 
> maintains dialog
>  >          state and must participate in all requests sent 
> on the dialogs
>  >          it has established.  
>  >
>  > We could of course say something like:
>  >
>  > "UAS, when part of a B2BUA"
> 
> according to the B2BUA definition:
> 
> 	Since it is a concatenation of a UAC and
>       UAS, no explicit definitions are needed for its behavior.
> 
> this means that there is no need to say anything special 
> about 199 either.  b2bus in terms of protocol specification, 
> is just an uas like any other.
> 
> -- juha
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use 
> sip-implementors@cs.columbia.edu for questions on current sip 
> Use sipping@ietf.org for new developments on the application of sip
> 
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
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	"Juha Heinanen" <jh@tutpro.com>
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Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
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Hi, 

>As you can never tell whether a UA is a UA by itself, or part of a
B2BUA, this must remain general UA behaviour.
>
>The most you can do is add some informative material that essentially
says this is useful for B2BUAs only.

My proposal was to simply talk about a UA which creates multiple To
tags. 

Regards,

Christer




> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of 
> Juha Heinanen
> Sent: Saturday, June 14, 2008 2:21 PM
> To: Christer Holmberg
> Cc: sip@ietf.org; Paul Kyzivat; Francois Audet
> Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
> 
> Christer Holmberg writes:
> 
>  >  Back-to-Back User Agent: A back-to-back user agent (B2BUA) is a
>  >          logical entity that receives a request and 
> processes it as a
>  >          user agent server (UAS).  In order to determine 
> how the request
>  >          should be answered, it acts as a user agent 
> client (UAC) and
>  >          generates requests.  Unlike a proxy server, it 
> maintains dialog
>  >          state and must participate in all requests sent 
> on the dialogs
>  >          it has established.  
>  >
>  > We could of course say something like:
>  >
>  > "UAS, when part of a B2BUA"
> 
> according to the B2BUA definition:
> 
> 	Since it is a concatenation of a UAC and
>       UAS, no explicit definitions are needed for its behavior.
> 
> this means that there is no need to say anything special about 199 
> either.  b2bus in terms of protocol specification, is just an uas like

> any other.
> 
> -- juha
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use 
> sip-implementors@cs.columbia.edu for questions on current sip Use 
> sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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From: "Rockson Li (zhengyli)" <zhengyli@cisco.com>
To: "Paul Kyzivat (pkyzivat)" <pkyzivat@cisco.com>,
	"Christer Holmberg" <christer.holmberg@ericsson.com>
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Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt
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The issue here is UAS may send 199 to terminate early-dialog with the inten=
tion to help UAC claiming resource
May bring unexpected overhead (case 1 && 2 as in Paul's mail) or confusion =
(case 1 in Paul's mail, ordered to terminate non-existed early-dialog)

Thanks

Rockson


-----Original Message-----
From: Paul Kyzivat (pkyzivat) =

Sent: Monday, June 16, 2008 8:18 PM
To: Christer Holmberg
Cc: Rockson Li (zhengyli); sip@ietf.org
Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt

I don't understand the problem here.
If the UAC receives the 199 as the first provisional response in a new dial=
og, then it will in effect create a new dialog and then immediately termina=
te it based on the 199.

If the UAC doesn't understand the new semantics of 199 then it will treat i=
t as an arbitrary 1xx response, and create a new dialog that then will have=
 to time out. But that is no different than if the 18x response had not bee=
n lost.

	Paul

Christer Holmberg wrote:
> Hi Rockson,
>  =

> Ok, now I understand. You are talking about the case where the UAC =

> doesn't receive the 18x, and then gets confused when it receives the =

> 199. That can happen no matter whether it's a proxy or UAS that sends =

> the 199.
>  =

> I guess that is something which could happen, yes, and a UAC could =

> think a new early dialog has been created. As you say, a way to avoid =

> it would be to send at least one of the 18x reliable before a 199 is sent.
>  =

> But, I am not sure we want to mandate that, because I don't think =

> anything will break if this happens.
>  =

> Regards,
>  =

> Chrsiter
>  =

>  =

> =

> ----------------------------------------------------------------------
> --
> *From:* Rockson Li (zhengyli) [mailto:zhengyli@cisco.com]
> *Sent:* 16. kes=E4kuuta 2008 4:51
> *To:* Christer Holmberg; sip@ietf.org
> *Subject:* Re: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt
> =

> Hi Christer,
>  =

> It would be easier for UAS to tell if it sends non-100 1xx response =

> creating early-dialog before sending 199.
> However, it's a little bit difficult from UAC's perspective, since the =

> non-100 1xx response might not be reliable.
>  =

> If UAC does not get , say 180, but get 199 first, it looks like it =

> does not make any sense.
>  =

> One option is to mandate usage of 100rel, which might be too strict =

> for this draft.
>  =

> Regards,
> -Rockson
> ----------------------------------------------------------------------
> ---------------------
> Date: Sun, 15 Jun 2008 13:52:24 +0200
> From: "Christer Holmberg" <christer.holmberg@ericsson.com =

> <mailto:christer.holmberg@ericsson.com>>
> Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt
> To: "Rockson Zhengyu Li" <lzyiii@laszlomail.com =

> <mailto:lzyiii@laszlomail.com>>, <sip@ietf.org <mailto:sip@ietf.org>>
> Message-ID:
>  =

> <CA9998CD4A020D418654FCDEF4E707DF06B19A2E@esealmw113.eemea.ericsson.se
> <mailto:CA9998CD4A020D418654FCDEF4E707DF06B19A2E@esealmw113.eemea.eric
> sson.se>>
>  =

> Content-Type: text/plain; charset=3D"iso-8859-1"
>  =

> Hi,
>  =

> You wouldn't send a 199 unless you have already sent a non-100 1xx =

> response, so the early dialog has already been created.
>  =

> Regards,
>  =

> Christer
>  =

> ________________________________
>  =

> From: sip-bounces@ietf.org <mailto:sip-bounces@ietf.org> =

> [mailto:sip-bounces@ietf.org] On Behalf Of Rockson Zhengyu Li
> Sent: 15. kes?kuuta 2008 14:43
> To: sip@ietf.org <mailto:sip@ietf.org>
> Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt
>  =

> =

> Hi,
>  =

> =

> one questions on this draft.
>  =

> =

> Since RFC3261 says, non-100 1xx response for INVITE cause early-dialog =

> creation.
> does this draft will normativly change the criteria, excluding 199?
>  =

> =

> If, the answer is Yes, I think it need be stated explicitly in the draft.
>  =

>  =

>  =

> thanks
>  =

> =

> Regards,
> -Rockson
>  =

>  =

> =

> =

> ----------------------------------------------------------------------
> --
> =

> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use =

> sip-implementors@cs.columbia.edu for questions on current sip Use =

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


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Hi,

having reading draft-ietf-sipping-nat-scenarios-08,
I got one question on sec 3.2.1.  Symmetric RTP/RTCP.

In this section, it says Symmetric RTP/RTCP is used to handle media 
tranversal issue in sec2,
but how does this be achieved?
I think Symmetric RTP/RTCP has nothing to do with addr/port 
advertisement in SDP,
addr/port will still be the private addr/port behind NAT even with 
Symmetric RTP/RTCP.

And why it should be used ONLY when one participant is in public 
network,
I think if NAT is of type Endpoint-Independent Mapping as per RFC4787, 
it should look fine.

Thanks

Regards,
-Rockson
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Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
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sreerekha.shenoy@wipro.com wrote:
> Thanks Christer,
> We have a query on our basic understanding of this draft.
>  =

> I too had the same query as Ashish, I can rephrase it for all of us.
> What we mean to ask is that, considering the scenario shown by the SIP =

> msg flow below, *at what point *does the proxy decide to send 199 =

> response *and on which *call legs.

I believe the time to send a 199 is when the proxy has received a final =

error response and has elected *not* to forward it on at that time - in =

hopes of getting a better response.

The 199 should be sent upstream using the to-tag from the final response =

that is not being forwarded.

> For e.g. in the scenario below, when proxy receives 403 on leg 2, and it =

> is yet to receive responses on leg 3 and 4. In this case, the proxy =

> cannot send a 199 to the UAC (is my understanding, please correct if wron=
g).

IMO you are wrong. What makes you think this isn't ok?

> Also after having received 5xx and 6xx on the leg 3 and 4, there is no =

> use sending 199 now because the best final response sent to UAC is =

> sufficient to indicate that the leg 3 and 4 are terminated.

At the time the 5xx is received there is still another possibility, so =

the 199 is appropriate.

When the 6xx is received, there are no more legs, and presumably the 6xx =

prevents the proxy from generating any more legs. It should be sending a =

final response immediately at that time, so I see no value in a 199 there.

BUT if the proxy takes a lenient view of how to handle 6xx (treating it =

as advisory rather than definitive regarding *global* failure) then it =

might decide that it now will be trying another destination. If so, the =

199 response might be appropriate. But the call flow is then incomplete.

	Thanks,
	Paul

> Please correct us if we are missing some hidden lines in the draft w.r.t =

> our understanding.
>  =

> Thanks,
> Best Regards,
> Rekha
> (ESN : 6-877-5834)
> (PSTN : 81168)
>  =

> =

> ------------------------------------------------------------------------
> *From:* sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] *On Behalf Of =

> *Christer Holmberg
> *Sent:* Monday, June 16, 2008 1:42 PM
> *To:* Ashish Saxena (WT01 - Telecom Equipment); pkyzivat@cisco.com; =

> Dale.Worley@comcast.net
> *Cc:* sip@ietf.org
> *Subject:* Re: [Sip] Draft submission: draft-ietf-sip-199-00
> =

>  =

> Hi,
>  =

> The way the "best" final response code is chosen is described in RFC =

> 3261, and the intention of 199 is not to change that.
>  =

> Regarding whether a 199 will be sent for leg 4: if the proxy is now =

> going to send a "best" final response I guess it doesn't need to send =

> 199 for leg 4.
>  =

> Regards,
>  =

> Christer
>  =

>  =

>  =

> =

> ------------------------------------------------------------------------
> *From:* sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] *On Behalf Of =

> *ashish.hsaxena@wipro.com
> *Sent:* 16. kes=E4kuuta 2008 10:32
> *To:* pkyzivat@cisco.com; Dale.Worley@comcast.net
> *Cc:* sip@ietf.org
> *Subject:* Re: [Sip] Draft submission: draft-ietf-sip-199-00
> =

> Referring to introduction part in the draft.
>    " At some point the proxy uses a specified mechanism to determine the =

> "best" final response code, and forwards that final response upstream =

> towards the sender of the associated request."
>  =

> Considering a scenario where all the forks have failed in yielding a 2xx =

> response from different UASes (UAS2 , 3 & 4 in the diagram) with =

> different error codes, how will the proxy determine the "best" final =

> response and to which leg would it send the best final response. What =

> would happen after 199 to leg 4?
>  =

>       UAC               P1                  UAS_2    UAS_3    UAS_4
> =

>        --- INVITE ------>
>                          --- INVITE (leg 2) ->
>                          --- INVITE (leg 3) ---------->
>                          --- INVITE (leg 4) ------------------->
>                          <-- 18x (leg 2) -----
>        <-- 18x (leg 2) --
>                          <-- 18x (leg 3) --------------
>        <-- 18x (leg 3) --
>                          <-- 18x (leg 4) -----------------------
>        <-- 18x (leg 4) --
>                          <-- *4xx* (leg 2) -----
>        <-- 199 (leg 2) --
>                          <-- *5xx *(leg 3) --------------
>        <-- 199 (leg 3) --
>                          <-- *6xx *(leg 4) -----------------------
>        <-- 199 (leg 4) --
> =

> Best Regards,
> Ashish Saxena
> 877-5570
> =

>  =

> =

> ------------------------------------------------------------------------
> *From:* sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] *On Behalf Of =

> *Attila Sipos
> *Sent:* Saturday, June 14, 2008 9:30 PM
> *To:* Paul Kyzivat; Dale.Worley@comcast.net
> *Cc:* sip@ietf.org
> *Subject:* Re: [Sip] Draft submission: draft-ietf-sip-199-00
> =

> =

>  >>However I am still scratching my head about the UAS sending a 199. Whi=
le
>  >>I see no reason to forbid it I'm trying to find a situation when it
>  >>would be helpful.
> =

> It may or may not be helpful but as you say, you can
> see no reason to forbid it.
> =

> Why restrict it if the restriction isn't needed?
> =

> As someone has pointed out, a B2BUA scenario might benefit
> from 199.  In fact, any kind of SIP gateway might benefit.
> =

>  > This is one of those cases that could probably benefit from
>  > simulation to judge the merits. My gut feel is that the costs
>  > of the UAS sending the 199 outweigh the benefits.
> =

> If the UAS wants to send it, that is a decision for
> them - not something that needs to be prevented.
> =

> SIP forking, in the technical sense of sending a
> response with a different to-tag, is always legal.
> So, any UA which has used forking
> should also be able to use 199.
> =

> I don't think we need to include any specific text about UAS
> except to say that a UAS can use it if the UAS has created
> the forking.
> =

> Regards,
> =

> Attila
> =

> =

> =

> -----Original Message-----
> From: sip-bounces@ietf.org on behalf of Paul Kyzivat
> Sent: Fri 13/06/2008 15:07
> To: Dale.Worley@comcast.net
> Cc: sip@ietf.org
> Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
> =

> This all sounds like good stuff.
> =

> However I am still scratching my head about the UAS sending a 199. While
> I see no reason to forbid it I'm trying to find a situation when it
> would be helpful. The only case I have been able to think of is:
> =

>    There is a forking proxy upstream of the UAS, but it doesn't
>    support 199. Yet it may delay the final response while it
>    tries other forks. So the UAS sending the 199 makes up for
>    the inadequacy of the forking proxy.
> =

> The trouble with this is that it is inefficient in many cases. If each
> proxy that forks also supports the sending of 199, and the UAS doesn't
> send one, then we avoid sending the message except in cases where it is
> helpful. If the UAS always sends it we have increased the message count
> by one for all failing calls. And the majority of failing calls get no
> benefit from a 199.
> =

> This is one of those cases that could probably benefit from simulation
> to judge the merits. My gut feel is that the costs of the UAS sending
> the 199 outweigh the benefits. Barring some analysis to the contrary I
> think I would say that a UAS SHOULD NOT send a 199. Or perhaps MAY send,
> but without any guidance of when.
> =

>         Paul
> =

> Dale.Worley@comcast.net wrote:
>  > Here are some comments on this I-D.  I don't think that any of them
>  > modify our understanding of the proposal.  I guess this is very terse;
>  > please use these suggestions as you see fit.
>  >
>  > * Abstract
>  >
>  > Clarify:
>  >
>  >    This specification defines a new SIP response code, 199 Early
>  >    Dialog Terminated, which a SIP entity can use to indicate upstream
>  >    that an early dialog has been terminated.  The response code can be
>  >    used by a SIP entity (a SIP proxy or UAS), to inform the UAC that
>  >    an early dialog has been terminated.  As the 199 response is a
>  >    provisional response, often the UAC will receive it before
>  >    receiving a final response to its request.
>  >
>  > * Section 1
>  >
>  >    When a forking proxy receives a non-2xx final response, it does not
>  >    always immediately forward the response upstream towards the sender
>  >    of the associated request.  Instead, the forking proxy "stores" it
>  >    and waits for further final responses from remote destinations where
>  >    the forked request was forwarded.
>  >
>  > The text does not allow for serial forking, only parallel forking,
>  > which might be confusing to people who are more aware of serial
>  > forking.  So amend the last sentence to:
>  >
>  >    Instead, the forking proxy "stores" it and waits for further final
>  >    responses from remote destinations where the forked request was
>  >    forwarded, and possibly forwards the request to additional
>  >    destinations.
>  >
>  > Also, Avoid using "..." except when indicating that the word is used
>  > in a non-standard way or that one is discussing the word itself:
>  >
>  >    Instead, the forking proxy "stores" it [...]
>  >
>  > should be:
>  >
>  >    Instead, the forking proxy stores it [...]
>  >
>  > * Section 3.
>  >
>  >    REQ 1: It must be possible to indicate to the UAC that an early
>  >    dialog has been terminated before a final response is sent.
>  >
>  > Oddly, though everyone thinks they understand this sentence, it
>  > doesn't really say what we want.  To really ensure that the UAC
>  > understands the early dialog has been terminated before the UAS sends
>  > a final response, the UAS would have to send 199 *and get the PRACK
>  > saying that it was received* before sending a final response.  This may
>  > well be worse than the present state of affairs.
>  >
>  > What we really mean is:
>  >
>  >    REQ 1: It must be possible for a UAS or proxy to send an indication
>  >    to the UAC that an early dialog has been terminated, which
>  >    indication will not be delayed waiting for a final response from
>  >    any UAS.
>  >
>  > * Section 4
>  >
>  >    When a client receives a 199 Early Dialog Terminated provisional
>  >    response it MAY release resources and procedures associated with the
>  >    early dialog the 199 response is received on.
>  >
>  > Amend to clarify the meaning of 199 independently of the consequences
>  > of that meaning:
>  >
>  >    When a client receives a 199 Early Dialog Terminated provisional
>  >    response, it has positive knowledge that the UAS that created the
>  >    early dialog has terminated it, and so the client MAY release
>  >    resources and procedures associated with the early dialog for which
>  >    the 199 response is received.
>  >
>  > * Section 4
>  >
>  > Add this paragraph before the last paragraph:
>  >
>  >    A UAC that receives a 199 response for an early dialog MUST NOT
>  >    send any further requests on that dialog, except for a PRACK if the
>  >    199 response was sent reliably, and PRACKs for other reliable
>  >    provisional responses that are to be acknowledged.
>  >
>  > This is copied from section 7 (Backward Compatibility), but since it
>  > is a constraint on UACs, it should also be stated in this section.
>  >
>  > * Section 4
>  >
>  >    The UAC MAY use additional information (for example the final
>  >    response which triggered the 199 response) received in the 199
>  >    response when initiating new sessions, if it is possible to avoid t=
he
>  >    new session initiation request to be rejected.
>  >
>  > What is the meaning of the final clause?
>  >
>  > * Section 5
>  >
>  > Add "early" as marked:
>  >
>  >    When a UAS wants to terminate *an early* dialog it sends a non-2xx
>  >    SIP final response, as specified in [RFC3261].  In addition, prior
>  >    to sending the non-2xx SIP final response, the UAS MAY send a 199
>  >    response to indicate that the dialog is being terminated.
>  >
>  > This clarifies that 199 cannot be used to terminate a confirmed
>  > dialog.
>  >
>  > (The reason a UAS may want to send a 199 is to provide this
>  > information even when the upstream proxy does not support 199.)
>  >
>  > There is a subtlty to the second sentence:
>  >
>  > Sending the 199 *before* the final response is not necessary due to
>  > the semantics of 199 (as it will probably reach the UAC first anyway,
>  > and it would not matter if it did not), but rather to prevent the
>  > upstream proxy from generating a 199 based on the final response, and
>  > then to transmit the UAS's 199 immediately after -- if the UAS sends
>  > the 199 first, any upstream proxy will (most likely) see the 199
>  > before seeing the final response (which might induce the proxy to send
>  > its own 199).
>  >
>  > * Section 6
>  >
>  >    When a forking proxy receives a non-2xx final response which
>  >    terminates one or more early dialogs and the proxy does not intend =
to
>  >    forward the final response immediately (due to the rules for a
>  >    forking proxy) it MUST send a 199 provisional response, for each
>  >    associated early dialog that it can associate with the final
>  >    response, upstream towards the sender of the associated request.
>  >
>  > Clarify, and note that if the proxy has already passed a 199 for an
>  > early dialog, it need not generate one itself.  Also, I've reduced
>  > MUST to SHOULD, because (as the following NOTE explains), the proxy
>  > may not know of all early dialogs which are terminated by the final
>  > response.
>  >
>  >    [...] it SHOULD initiate a 199 provisional response upstream
>  >    towards the sender of the associated request, for each terminated
>  >    early dialog, excepting those for which it has already transmitted
>  >    a 199 response.
>  >
>  > Note that the "already transmitted 199 responses" may have come from
>  > downstream proxies as well as downstream UASs, so this consideration
>  > applies even if we do not allow UASs to send 199s.
>  >
>  > * Section 7
>  >
>  >    Since all SIP entities involved in a session setup do not necessari=
ly
>  >    support the specific meaning of the 199 Early Dialog Terminated
>  >    provisional response, the sender of the response MUST be prepared to
>  >    receive SIP requests associated with the dialog for which the 199
>  >    response was sent.
>  >
>  > This situation can also result due to race conditions.  Amend to:
>  >
>  >    Because of race conditions and because the client may not support t=
he
>  >    specific meaning of the 199 provisional response, the sender of the
>  >    response MUST be prepared to receive SIP requests associated with
>  >    the dialog for which the 199 response was sent.
>  >
>  > And add a qualifier to this sentence:
>  >
>  >    If such request is received, and the receiver maintains state of
>  >    the dialogs, the receiver MUST reply to such requests with a 481
>  >    final response (unless another error response is more appropriate).
>  >
>  > * Section 7
>  >
>  > Remove "..." from this sentence and add "even" as marked:
>  >
>  >    The 199 Early Dialog Terminated response code MUST NOT replace a
>  >    final response.  A final response MUST always be sent, *even* after
>  >    one or many 199 Early Dialog Terminated provisional responses have
>  >    been sent.
>  >
>  > * Section 8
>  >
>  > Clarify this sentence to:
>  >
>  >    The 199 Early Dialog Terminated response code allows a SIP entity
>  >    to indicate to upstream entities that a specific early dialog has
>  >    been terminated, before a final response reaches the upstream
>  >    entities.
>  >
>  > Add the following:
>  >
>  >    The early dialog that has been terminated is identified by the Call=
-Id
>  >    header and the to-tag value of the 199 response.
>  >
>  >    Future standards may define optional additional information carried
>  >    in the headers and/or body of the response.
>  >
>  > * Section 9
>  >
>  >    A 199 Early Dialog Terminated provisional response MUST NOT contain=
 a
>  >    new SDP offer/answer message body, but the sender of the response M=
AY
>  >    insert a copy of a previously sent offer/answer message body as
>  >    otherwise allowed by the offer/answer rules for a provisional
>  >    response.
>  >
>  > This rule seems excessively strict, as it seems OK to send a new SDP
>  > answer in a 199, even if the dialog will be immediately terminated.
>  > Certainly, the UAC cannot tell the difference, as there could have
>  > been a previous but lost 1xx response carrying the same SDP.
>  >
>  > The interaction of 199 and SDP probably needs further study...
>  >
>  > * Section 11
>  >
>  > Nit:  Use "reject" here rather than "rejects":
>  >
>  >    UAS_2 and UAS_3 *reject* the INVITE by sending a 4xx error response.
>  >
>  > Here is a modification of the example, showing how a UAS can send a
>  > 199, and how the proxy handles it.  (You may or may not want to use
>  > this.)  I've also added the ACK's for the 4xx's, which were missing in
>  > the original.
>  >
>  >    [...] UAS_2 and UAS_3 reject the INVITE by sending a 4xx error
>  >    response.  UAS_3 sends a 199 with its 4xx.  When P1 receives the
>  >    4xx response from UAS_2 it immediately sends a 199 Early Dialog
>  >    Terminated associated with UAS_2's dialog towards the UAC.  P1
>  >    transmits UAS_3's 199, and so does not initiate one itself for
>  >    UAS_3's dialog when it receives UAS_3's 4xx response.
>  >
>  >       UAC               P1                  UAS_2    UAS_3    UAS_4
>  >
>  >        --- INVITE ------>
>  >                          --- INVITE (leg 2) ->
>  >                          --- INVITE (leg 3) ---------->
>  >                          --- INVITE (leg 4) ------------------->
>  >                          <-- 18x (leg 2) -----
>  >        <-- 18x (leg 2) --
>  >                          <-- 18x (leg 3) --------------
>  >        <-- 18x (leg 3) --
>  >                          <-- 18x (leg 4) -----------------------
>  >        <-- 18x (leg 4) --
>  >                          <-- 4xx (leg 2) -----
>  >        <-- 199 (leg 2) --
>  >                          --- ACK (leg 2) ---->
>  >                          <-- 199 (leg 3) --------------
>  >        <-- 199 (leg 3) --
>  >                          <-- 4xx (leg 3) --------------
>  >                          --- ACK (leg 3) ------------->
>  >                          <-- 200 (leg 4) -----------------------
>  >        <-- 200 (leg 4) --
>  >        --- ACK (leg 4) ->
>  >                          --- ACK (leg 4) ---------------------->
>  >
>  > Dale
>  > _______________________________________________
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>  > This list is for NEW development of the core SIP Protocol
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>  > Use sipping@ietf.org for new developments on the application of sip
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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-15.txt
	Pages		: 48
	Date		: 2008-6-16
	
The Session Initiation Protocol (SIP) allows proxy servers to
   initiate TCP connections or to send asynchronous UDP datagrams to
   User Agents in order to deliver requests.  However, in a large number
   of real deployments, many practical considerations, such as the
   existence of firewalls and Network Address Translators (NATs) or the
   use of TLS with server-provided certificates, 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 from
   the User Agent to its Registrar.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-outbound-15.txt

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ftp://ftp.ietf.org/internet-drafts/

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_______________________________________________
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From sip-bounces@ietf.org  Mon Jun 16 16:22:48 2008
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Subject: [Sip] Toward the Evolution of SIP and Related Working Groups
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The SIP working group spun off from the MMUSIC working group when we  
realized that the SIP work was reasonably separable from the other  
work going on in MMUSIC.

The SIP working group first met officially at IETF 46 in November  
1999, although we did have an earlier interim meeting.

This means that the fall 1999 meeting will be our 10th anniversary. In  
addition to having a big party, I'd also like to plan for this being  
the last meeting of the SIP working group as it is now defined. Ten  
years is long enough. It's time we restructured.

There will probably still be people who want to change, revise,  
extend, or otherwise manipulate the family of SIP specifications after  
2009.

So what should we do? How should we organize to get those needs met?   
Rest assured, we'll be having further discussions on this topic in the  
coming months. But I'd like to start the discussion now.

By November 2009, I fully expect SIP to have completed the majority of  
currently chartered items. Between now and then, we'll probably also  
take on and complete a bit of new work (although I plan to be very  
resistant to new work). I also expect that there are a few items on  
our charter now that we're not going to finish. It may well be that  
some of those items need a different organizational structure to  
progress than what we have. Or maybe they don't need doing.

Further, I've come to believe that most working groups should be  
shorter-lived and tightly constrained by their charters to produce a  
single concise set of deliverables. There's still a need for a longer- 
lived thing that maintains architectural oversight. I've come to  
believe that this is the "area", not the working group.

Let's start the recap by looking at the current charter of the SIP WG.  
I've eliminated the WGLC milestones to shorten the list, as each has a  
following IESG milestone. Note that there's a Feb 2008 milestone for  
"Revise Charter". You can consider this topic the first installment on  
that milestone, even though it is a bit late.

1) Jul 2007 Location Conveyance with SIP to IESG (PS)

2) Aug 2007 Connection reuse mechanism to IESG (PS)

3) Sep 2007 Session Policies to IESG as PS

4) Sep 2007 Example security flows to IESG (Informational)

5) Oct 2007 New resource priority namespaces for DISA to IESG (PS)

6) Nov 2007 Requirements for media keying to IESG (Informational)

7 )Dec 2007 Extensions to SIP UA Profile Delivery Change Notification  
Event Package for XCAP to IESG (PS)

8) Dec 2007 Roadmap for SIP to IESG (Informational)

9) Dec 2007 Using SAML for SIP to IESG (PS)

10) Dec 2007 Extension for use in etags in conditional notification to  
IESG (PS)

11) Feb 2008 Identify requirements for test matrix to move SIP to  
Draft Standard

12) Feb 2008 Delivering request-URI and parameters to UAS via proxy to  
IESG (PS)

13) Feb 2008 Revise charter

14) Feb 2008 Establishment of secure media sessions using DTLS-SRTP to  
IESG (PS)

15) Apr 2008 MIME body handling in SIP to IESG (PS)

16) Apr 2008 Essential corrections to RFC 3261 (1st batch) to IESG (PS)

17) Jul 2008 Mechanisms for UA initiated privacy to IESG (BCP)

18) Aug 2008 Guidelines for double route recording to IESG (BCP)

19) Sep 2008 X.509 Certificates for TLS use in SIP to IESG (PS)

20) Sep 2008 X.509 extended key usage for SIP to IESG (PS)

21) Nov 2008 Termination of early dialog prior to final response to  
IESG (PS)


Looking at this, we have a couple of discrete tasks that I think might  
make the basis for different narrowly-scoped working groups. We've  
also got a lot of stuff that we're late on delivering!

1) SIP WG (to finish by end of 2009) gets the bulk of these items.

2) SIP Draft Standard WG: Takes the milestone of "Identify  
requirements for test matrix to move SIP to Draft Standard" and all of  
the follow-on work needed to move SIP from Proposed to Draft on the  
standards track. There's certainly enough work here for a dedicated  
working group, with the last milestone being something like "Deliver  
SIP as a Draft Standards Document to IESG". We may wish to consider  
whether we really want to go this route or whether we should instead  
be thinking of a "SIP 3.0" effort. My personal though is that there's  
just too much cruft in the RFC 3261 family to make Draft, but I could  
be proven wrong. I'd rather see us put SIP 2.0 into maintenance mode  
at the end of 2009 and move the bulk of our resources into developing  
a SIP 3.0 family designed from the ground up to be Draft-Standard  
achievable.

3) RAI Essential Corrections WG: Does this justify its own working  
group? Remember, it would be not just RFC 3261, but the entire family  
of related RFCs that would need lock-step maintenance and corrections.  
Another approach might be to roll the corrections into the Draft  
Standard version of SIP 2.0.

4) RAI Policies WG: We have these ongoing, not-quite-committed work  
items like session policies, use of SAML, etc.  that we've never been  
able to get fully wrapped up. I suspect they need their own WG. Or we  
need to decide that these things just aren't worth doing and stop  
feeling guilty about not finishing. As it is, I don't have a good  
feeling about SIP finishing them.

5) SIP Operations: I'd like to see SIPPING replaced with a "how to"  
group possibly under the OPS banner. They could deal with all the "how  
do we do X with SIP" from other SDOs. When confronted by a need to  
extend SIP to meet a given need, they could then BOF up a new WG to  
deal with the required extension.

We already have working groups looking at conferencing, peering,  
telephony feature interop, provisioning, peer-to-peer, NAT traversal,

Other things that might need WGs: SPIT? Identity and Reputation?

I also suspect we need a couple of directorates that are not WGs to  
act in an advisory capacity:

1) RAI Security Directorate: Focuses on security review and  
architectural issues in RAI documents.

2) RAI Event Package Directorate: Reviews and approves new SIP event  
packages




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Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
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   From: Paul Kyzivat <pkyzivat@cisco.com>

   Well, the specification could be abstracted to remove the mention of the 
   B2BUA while still having the same effect. It would have to say something 
   about a UAS should send a 199 if it knows the dialog will be terminated 
   before it can send the final response. But I think we would still need 
   to include an example of a B2BUA in order for anybody to have a clue 
   what we were talking about.

I think the situation is simpler than people are making it out to be.

The situations in which having an agent generate and send a 199 is
useful are when all of the following are true:

- the agent knows that an early dialog has been created because it
  transmitted a 1xx response (either that it generated, or that it
  received from downstream) (And as Rocksun pointed out, "1xx"
  explicitly excludes both "100" and "199".), and

- the agent knows that the early dialog has been terminated due to
  receiving a failure final response which includes the early dialog
  (which was from the downstream leg from which the 1xx came), and

- the agent has not transmitted a 199 response to terminate the early
  dialog, and

- the agent will not be immediately sending a failure final response
  (which would indicate the termination of the early dialog)

It turns out that an ordinary UAS never is in this situation (because
it only generates one early dialog for an INVITE, and only ends the
early dialog when it sends a failure final response), so by
implication, an ordinary UAS never should send a 199.

If we enumerate the above criteria, we don't have to describe the
various cases of UA vs. B2BUA, etc., because they're all covered.  And
by making the criteria explicit, we increase the probability of useful
and effective implementations.

Dale
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From: Paul Kyzivat <pkyzivat@cisco.com>
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Dale,

I'm fine with spinning it this way. It avoids using the *term* B2BUA, 
but it still is talking about a B2BUA since it is talking about a UA 
that has received a response from downstream of itself. So we are 
playing political correctness gavmes. But I'm ok with that.

	Thanks,
	Paul

Dale.Worley@comcast.net wrote:
>    From: Paul Kyzivat <pkyzivat@cisco.com>
> 
>    Well, the specification could be abstracted to remove the mention of the 
>    B2BUA while still having the same effect. It would have to say something 
>    about a UAS should send a 199 if it knows the dialog will be terminated 
>    before it can send the final response. But I think we would still need 
>    to include an example of a B2BUA in order for anybody to have a clue 
>    what we were talking about.
> 
> I think the situation is simpler than people are making it out to be.
> 
> The situations in which having an agent generate and send a 199 is
> useful are when all of the following are true:
> 
> - the agent knows that an early dialog has been created because it
>   transmitted a 1xx response (either that it generated, or that it
>   received from downstream) (And as Rocksun pointed out, "1xx"
>   explicitly excludes both "100" and "199".), and
> 
> - the agent knows that the early dialog has been terminated due to
>   receiving a failure final response which includes the early dialog
>   (which was from the downstream leg from which the 1xx came), and
> 
> - the agent has not transmitted a 199 response to terminate the early
>   dialog, and
> 
> - the agent will not be immediately sending a failure final response
>   (which would indicate the termination of the early dialog)
> 
> It turns out that an ordinary UAS never is in this situation (because
> it only generates one early dialog for an INVITE, and only ends the
> early dialog when it sends a failure final response), so by
> implication, an ordinary UAS never should send a 199.
> 
> If we enumerate the above criteria, we don't have to describe the
> various cases of UA vs. B2BUA, etc., because they're all covered.  And
> by making the criteria explicit, we increase the probability of useful
> and effective implementations.
> 
> Dale
> _______________________________________________
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> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
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> 
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On Jun 16, 2008, at 9:51 PM, Dale.Worley@comcast.net wrote:

>   From: Paul Kyzivat <pkyzivat@cisco.com>
>
>   Well, the specification could be abstracted to remove the mention  
> of the
>   B2BUA while still having the same effect. It would have to say  
> something
>   about a UAS should send a 199 if it knows the dialog will be  
> terminated
>   before it can send the final response. But I think we would still  
> need
>   to include an example of a B2BUA in order for anybody to have a clue
>   what we were talking about.

Could it be explained in terms of a UA that is acting as a protocol  
gateway?

--
Dean

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We have (theoretically) four hours in Dublin.

Let's recap our milestones and see what needs to be talked about in  
dublin:


1) Jul 2007 Location Conveyance with SIP to IESG (PS)
-- as far as I know, this is stable and the retransmission issue is  
being hashed in geopriv

2) Aug 2007 Connection reuse mechanism to IESG (PS)
-- I believe this is awaiting a proto writeup and that the WG is done  
with it

3) Sep 2007 Session Policies to IESG as PS
-- I believe this is awaiting a proto writeup and that the WG is done  
with it

4) Sep 2007 Example security flows to IESG (Informational)
-- Ok, I've lost track on this one.

5) Oct 2007 New resource priority namespaces for DISA to IESG (PS)
-- I believe this is awaiting a proto writeup and that the WG is done  
with it

6) Nov 2007 Requirements for media keying to IESG (Informational)
-- I believe this is awaiting a proto writeup and that the WG is done  
with it

7 )Dec 2007 Extensions to SIP UA Profile Delivery Change Notification
Event Package for XCAP to IESG (PS)
-- I believe this is awaiting a proto writeup and that the WG is done  
with it

8) Dec 2007 Roadmap for SIP to IESG (Informational)
-- I believe this is awaiting a proto writeup and that the WG is done  
with it

9) Dec 2007 Using SAML for SIP to IESG (PS)
-- We seem to be stalled. I've heard from one person who is excitedly  
implementing, so at least there's something like a constituency.

10) Dec 2007 Extension for use in etags in conditional notification to
IESG (PS)
-- we are waiting on the author to finish some changes from WGLC, and  
have no issues to discuss.

11) Feb 2008 Identify requirements for test matrix to move SIP to
Draft Standard
-- I guess we need to talk about this.

12) Feb 2008 Delivering request-URI and parameters to UAS via proxy to
IESG (PS)
-- we seem to be stuck on this one, with two conflicting  
methodologies. How can we structure a conversation to produce  
meaningful results?

13) Feb 2008 Revise charter
-- I've started this discussion on-list

14) Feb 2008 Establishment of secure media sessions using DTLS-SRTP to
IESG (PS)
-- We were last discussing the impact of RFC 4474's PSTN-interworking  
problem on this draft, and got stuck. I guess we should talk about it.

15) Apr 2008 MIME body handling in SIP to IESG (PS)
-- the latest version supposedly addressed all list comments. Where  
are we?

16) Apr 2008 Essential corrections to RFC 3261 (1st batch) to IESG (PS)
-- This is a perennial favorite for discussion, as Essential  
Corrections are essential confusing to a lot of people.

17) Jul 2008 Mechanisms for UA initiated privacy to IESG (BCP)
-- This seems to have gone forward as an individual draft.

18) Aug 2008 Guidelines for double route recording to IESG (BCP)
-- This seems to have stalled. Do we have any issues? I'm not aware of  
any

19) Sep 2008 X.509 Certificates for TLS use in SIP to IESG (PS)
-- We are awaiting a revised draft

20) Sep 2008 X.509 extended key usage for SIP to IESG (PS)
-- we are awaiting a revised draft

21) Nov 2008 Termination of early dialog prior to final response to
IESG (PS)
-- we seem to be discussing on-list.


This leaves the following probable topics:

1) Example Security Flows
2) SAML
3) Test Matrix for Draft Standard
4) Delivering Request-URI and Parameters to UAS via Proxy
5) DTLS-SRTP and RFC 4474
6) MIME Body Handling
7) Essential Corrections
8) Termination of Early Dialog Prior to Final Response

So, the floor is now open for discussing what should be on the agenda.  
Remember, chartered work (as listed above) gets first priority.

I'm also deeply concerned about #4 and #5 above and trying to figure  
out how to get traction. Anybody got an idea?

--
Dean, SIP chair for the last 9 years . . .



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> 4) Sep 2007 Example security flows to IESG (Informational)
> -- Ok, I've lost track on this one.
In my hands waiting for some information from ekr.

> 11) Feb 2008 Identify requirements for test matrix to move SIP to
> Draft Standard
> -- I guess we need to talk about this.
Agreed - I hope to have a strawman of some sort this time around.

> 16) Apr 2008 Essential corrections to RFC 3261 (1st batch) to IESG  
> (PS)
> -- This is a perennial favorite for discussion, as Essential  
> Corrections are essential confusing to a lot of people.
I will start another thread on this.

>
>
>
> This leaves the following probable topics:
>
> 1) Example Security Flows
This needs no agenda time.
>
> 2) SAML
> 3) Test Matrix for Draft Standard
A short amount of time for this (if substantiated by list discussion)  
would be useful.
>
> 4) Delivering Request-URI and Parameters to UAS via Proxy
> 5) DTLS-SRTP and RFC 4474
> 6) MIME Body Handling
> 7) Essential Corrections
Lets see how the other thread I start goes - hopefully we can actually  
do something here on list instead of eating face-time for it.
>
> 8) Termination of Early Dialog Prior to Final Response


I strongly agree that we should focus on chartered work, and suggest  
that only things really getting list attention take significant agenda  
time.

>
>
> So, the floor is now open for discussing what should be on the agenda.
> Remember, chartered work (as listed above) gets first priority.

Where is the overload discussion? (I'm not suggesting agenda time  
necessarily)


>
>
> I'm also deeply concerned about #4 and #5 above and trying to figure
> out how to get traction. Anybody got an idea?
>
> --
> Dean, SIP chair for the last 9 years . . .
>
>
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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From: Robert Sparks <rjsparks@nostrum.com>
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Subject: [Sip] Essential corrections checkpoint
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So, for essential corrections as we've talked about them so far:

fork-loop-fix is back in the chairs hands (for a long long time - can  
we escalate this? The version that needed to go back in was submitted  
in November - it's technically expired, but the "AD is Watching"  
designation is keeping it in the tracker).

invfix will be revised before the dublin -00. That version should be  
ready to LC

record-route-fix we took out of essential corrections - its moving  
forward as a BCP the last I saw traffic about it.

Am I forgetting about something else?

I still think fork-loop-fix should go through as a standalone  
document. That leaves only invfix to be part of the "first" bundle of  
things to go and I have no visibility into anything else that would go  
into that bundle, so, effectively, _it_ would be a standalone document.

RjS

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Robert Sparks wrote:
> So, for essential corrections as we've talked about them so far:
> 
> fork-loop-fix is back in the chairs hands (for a long long time - can  
> we escalate this? The version that needed to go back in was submitted  
> in November - it's technically expired, but the "AD is Watching"  
> designation is keeping it in the tracker).
> 
> invfix will be revised before the dublin -00. That version should be  
> ready to LC
> 
> record-route-fix we took out of essential corrections - its moving  
> forward as a BCP the last I saw traffic about it.
> 
> Am I forgetting about something else?

Robert: IPv6 ABNF fix draft (see
http://tools.ietf.org/html/draft-ietf-sip-ipv6-abnf-fix-02).  The
last known and stable version is -02 and ready to be moved forward.

> I still think fork-loop-fix should go through as a standalone  
> document. That leaves only invfix to be part of the "first" bundle of  
> things to go and I have no visibility into anything else that would go  
> into that bundle, so, effectively, _it_ would be a standalone document.

I believe the ipv6-abnf-fix was scheduled to go in the first bundle.

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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Thanks Vijay -

That's why I asked what I was forgetting.

RjS

On Jun 17, 2008, at 10:10 AM, Vijay K. Gurbani wrote:

> Robert Sparks wrote:
>> So, for essential corrections as we've talked about them so far:
>> fork-loop-fix is back in the chairs hands (for a long long time -  
>> can  we escalate this? The version that needed to go back in was  
>> submitted  in November - it's technically expired, but the "AD is  
>> Watching"  designation is keeping it in the tracker).
>> invfix will be revised before the dublin -00. That version should  
>> be  ready to LC
>> record-route-fix we took out of essential corrections - its moving   
>> forward as a BCP the last I saw traffic about it.
>> Am I forgetting about something else?
>
> Robert: IPv6 ABNF fix draft (see
> http://tools.ietf.org/html/draft-ietf-sip-ipv6-abnf-fix-02).  The
> last known and stable version is -02 and ready to be moved forward.
>
>> I still think fork-loop-fix should go through as a standalone   
>> document. That leaves only invfix to be part of the "first" bundle  
>> of  things to go and I have no visibility into anything else that  
>> would go  into that bundle, so, effectively, _it_ would be a  
>> standalone document.
>
> I believe the ipv6-abnf-fix was scheduled to go in the first bundle.
>
> 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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One question and one comment below [MB].

Mary.

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Robert Sparks
Sent: Tuesday, June 17, 2008 9:29 AM
To: Dean Willis
Cc: sip@ietf.org
Subject: Re: [Sip] WG Status, and Towards a Dublin SIP Agenda

> 4) Sep 2007 Example security flows to IESG (Informational)
> -- Ok, I've lost track on this one.
In my hands waiting for some information from ekr.

> 11) Feb 2008 Identify requirements for test matrix to move SIP to 
> Draft Standard
> -- I guess we need to talk about this.
Agreed - I hope to have a strawman of some sort this time around.

> 16) Apr 2008 Essential corrections to RFC 3261 (1st batch) to IESG
> (PS)
> -- This is a perennial favorite for discussion, as Essential 
> Corrections are essential confusing to a lot of people.
I will start another thread on this.

>
>
>
> This leaves the following probable topics:
>
> 1) Example Security Flows
This needs no agenda time.
>
> 2) SAML

[MB] What more needs to be done in this area? I thought the document was
fairly complete or are there outstanding issues? [/MB]

> 3) Test Matrix for Draft Standard
A short amount of time for this (if substantiated by list discussion)
would be useful.
>
> 4) Delivering Request-URI and Parameters to UAS via Proxy
> 5) DTLS-SRTP and RFC 4474
> 6) MIME Body Handling
> 7) Essential Corrections
Lets see how the other thread I start goes - hopefully we can actually
do something here on list instead of eating face-time for it.
>
> 8) Termination of Early Dialog Prior to Final Response


I strongly agree that we should focus on chartered work, and suggest
that only things really getting list attention take significant agenda
time.

>
>
> So, the floor is now open for discussing what should be on the agenda.
> Remember, chartered work (as listed above) gets first priority.

Where is the overload discussion? (I'm not suggesting agenda time
necessarily)

[MB] We should allow agenda time in SIPPING for an update/status from
the Overload design team. The SIPPING chairs (I)  still have an
outstanding action to define a WG deliverable for the output of the
design team. Volker will be splitting the solution out of his current
individual document and discuss this as a possible document to meet that
deliverable. Whether we progress that document or merge it back with the
solution later should be considered. But, for now, we should keep
discussion of this topic in SIPPING WG. [/MB]

>
>
> I'm also deeply concerned about #4 and #5 above and trying to figure 
> out how to get traction. Anybody got an idea?
>
> --
> Dean, SIP chair for the last 9 years . . .
>
>
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use 
> sip-implementors@cs.columbia.edu for questions on current sip Use 
> sipping@ietf.org for new developments on the application of sip

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Thread-Topic: [Sip] Toward the Evolution of SIP and Related Working Groups
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Dean Willis" <dean.willis@softarmor.com>, <sip@ietf.org>
Subject: Re: [Sip] Toward the Evolution of SIP and Related Working Groups
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I have a few comments below [MB].  And one more mega comment/proposal
here.  Mobile IP is an area that has a similar history as SIP in terms
of the demand and explosion of work. They considered our SIP/SIPPING
model (at that time we had few other spinoffs, as I recall) when they
split their WGs. They have a group for IPv6 work(mext), one for
IPv4(mipv4) and one for performance, signaling and
optimizations(mipshop). So, from you list of numbers, I could also see a
3 WG split for SIP/SIPPING (discounting all the current spinoffs) using
your #s (and assuming SIP WG is done):
A) #2+#3: SIP core protocol and extensions to fold into next protocol
version
B) #4:  Policies and security related aspects of SIP Protocol and then
put the burden of screening new work such as SPIT, identity, etc. in
this group until it's decided that the workload demands a separate WG.  
C) #5: Basically, SIPPING slightly reduced from current charter  -
filtering  (including OPS related items), howtos, discussing p-headers,
etc. - the primary difference here is that this group would not screen
basic SIP extension proposals or any of the policy/security, etc.
related work. Right now we don't do the security, but some of the other
stuff has been debatable.  

In my mind, the primary thing this would accomplish would be separating
out the policy and security work to ensure there's the right focus and
attention. I also think this makes scheduling only slightly easier in
that it might be easier to ensure security folks could attend. 

However, I don't think this really helps that much with overall workload
- it's just moving some things around. For example,  if A above does all
the SIP extension screening, that workload may make A be just as crazy
as SIP is now OR if in the end the security stuff is causing the
craziness, then group B becomes the crazy one, until it sheds more work
to new WGs and of course C stays fairly sane, which is what I think it
is now ;)  

In general, I agree that SIP should follow the model that SIPPING did a
couple years ago and NOT accept new work items until currently chartered
items are done (or at least down to a manageable handful (5-6)) rather
than the current 17 (use the tools page rather than WG charter to see
the delta), so putting the WG on a path to closure is an excellent idea.


Mary.

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Dean Willis
Sent: Monday, June 16, 2008 6:23 PM
To: sip@ietf.org
Subject: [Sip] Toward the Evolution of SIP and Related Working Groups


The SIP working group spun off from the MMUSIC working group when we
realized that the SIP work was reasonably separable from the other work
going on in MMUSIC.

The SIP working group first met officially at IETF 46 in November 1999,
although we did have an earlier interim meeting.

This means that the fall 1999 meeting will be our 10th anniversary. In
addition to having a big party, I'd also like to plan for this being the
last meeting of the SIP working group as it is now defined. Ten years is
long enough. It's time we restructured.

There will probably still be people who want to change, revise, extend,
or otherwise manipulate the family of SIP specifications after 2009.

So what should we do? How should we organize to get those needs met?   
Rest assured, we'll be having further discussions on this topic in the
coming months. But I'd like to start the discussion now.

By November 2009, I fully expect SIP to have completed the majority of
currently chartered items. Between now and then, we'll probably also
take on and complete a bit of new work (although I plan to be very
resistant to new work). I also expect that there are a few items on our
charter now that we're not going to finish. It may well be that some of
those items need a different organizational structure to progress than
what we have. Or maybe they don't need doing.

Further, I've come to believe that most working groups should be
shorter-lived and tightly constrained by their charters to produce a
single concise set of deliverables. There's still a need for a longer-
lived thing that maintains architectural oversight. I've come to believe
that this is the "area", not the working group.

Let's start the recap by looking at the current charter of the SIP WG.  
I've eliminated the WGLC milestones to shorten the list, as each has a
following IESG milestone. Note that there's a Feb 2008 milestone for
"Revise Charter". You can consider this topic the first installment on
that milestone, even though it is a bit late.

1) Jul 2007 Location Conveyance with SIP to IESG (PS)

2) Aug 2007 Connection reuse mechanism to IESG (PS)

3) Sep 2007 Session Policies to IESG as PS

4) Sep 2007 Example security flows to IESG (Informational)

5) Oct 2007 New resource priority namespaces for DISA to IESG (PS)

6) Nov 2007 Requirements for media keying to IESG (Informational)

7 )Dec 2007 Extensions to SIP UA Profile Delivery Change Notification
Event Package for XCAP to IESG (PS)

8) Dec 2007 Roadmap for SIP to IESG (Informational)

9) Dec 2007 Using SAML for SIP to IESG (PS)

10) Dec 2007 Extension for use in etags in conditional notification to
IESG (PS)

11) Feb 2008 Identify requirements for test matrix to move SIP to Draft
Standard

12) Feb 2008 Delivering request-URI and parameters to UAS via proxy to
IESG (PS)

13) Feb 2008 Revise charter

14) Feb 2008 Establishment of secure media sessions using DTLS-SRTP to
IESG (PS)

15) Apr 2008 MIME body handling in SIP to IESG (PS)

16) Apr 2008 Essential corrections to RFC 3261 (1st batch) to IESG (PS)

17) Jul 2008 Mechanisms for UA initiated privacy to IESG (BCP)

18) Aug 2008 Guidelines for double route recording to IESG (BCP)

19) Sep 2008 X.509 Certificates for TLS use in SIP to IESG (PS)

20) Sep 2008 X.509 extended key usage for SIP to IESG (PS)

21) Nov 2008 Termination of early dialog prior to final response to IESG
(PS)


Looking at this, we have a couple of discrete tasks that I think might
make the basis for different narrowly-scoped working groups. We've also
got a lot of stuff that we're late on delivering!

1) SIP WG (to finish by end of 2009) gets the bulk of these items.

2) SIP Draft Standard WG: Takes the milestone of "Identify requirements
for test matrix to move SIP to Draft Standard" and all of the follow-on
work needed to move SIP from Proposed to Draft on the standards track.
There's certainly enough work here for a dedicated working group, with
the last milestone being something like "Deliver SIP as a Draft
Standards Document to IESG". We may wish to consider whether we really
want to go this route or whether we should instead be thinking of a "SIP
3.0" effort. My personal though is that there's just too much cruft in
the RFC 3261 family to make Draft, but I could be proven wrong. I'd
rather see us put SIP 2.0 into maintenance mode at the end of 2009 and
move the bulk of our resources into developing a SIP 3.0 family designed
from the ground up to be Draft-Standard achievable.
[MB] I like this suggestion. [/MB]

3) RAI Essential Corrections WG: Does this justify its own working
group? Remember, it would be not just RFC 3261, but the entire family of
related RFCs that would need lock-step maintenance and corrections.  
Another approach might be to roll the corrections into the Draft
Standard version of SIP 2.0.
[MB] I think this should be roleed into 2) above. [/MB]


4) RAI Policies WG: We have these ongoing, not-quite-committed work
items like session policies, use of SAML, etc.  that we've never been
able to get fully wrapped up. I suspect they need their own WG. Or we
need to decide that these things just aren't worth doing and stop
feeling guilty about not finishing. As it is, I don't have a good
feeling about SIP finishing them.
[MB] I like this idea and would it also not cover any outstanding
ongoing security items or are you anticipating that to addressed
entirely by the currently chartered work items?  [/MB]

5) SIP Operations: I'd like to see SIPPING replaced with a "how to"  
group possibly under the OPS banner. They could deal with all the "how
do we do X with SIP" from other SDOs. When confronted by a need to
extend SIP to meet a given need, they could then BOF up a new WG to deal
with the required extension.
[MB] Are you proposing this to replace SIPPING WG?  If so, I don't see
it falling under the banner of OPS. I do see that there may potentially
be the need to initially discuss OPS related items, but as in the past,
I see those being completed in the appropriate OPS WGs (e.g.,
performance).  I have mixed views on how effective it would be to have a
BoF and new WG for new extensions. I don't think the model we currently
have is that ineffective in terms of first discussing requirements
before agreeing a solution.  I see a SIP Maintenace group perhaps to
handle these items, although in many cases, I think it would be far more
effective to roll these things into your item 2 - i.e., don't you see
SIP 3.0 as pulling in a lot of the headers that have been defined since
RFC 3261 has been published. [/MB]

We already have working groups looking at conferencing, peering,
telephony feature interop, provisioning, peer-to-peer, NAT traversal,

Other things that might need WGs: SPIT? Identity and Reputation?

[MB] I would agree these two topics might warrant separate WGs, although
some of that work overlaps with your WG 4, I believe, the division of
work may not be so clear.  [/MB]

I also suspect we need a couple of directorates that are not WGs to act
in an advisory capacity:

[MB] I think we already have these and more covered by our current RAI
area review team. These are specialized reviews and assignment of
reviewers is based on area of expertise and we've got security and event
package gurus already on the team. [/MB]

1) RAI Security Directorate: Focuses on security review and
architectural issues in RAI documents.

2) RAI Event Package Directorate: Reviews and approves new SIP event
packages




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Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
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   From: Paul Kyzivat <pkyzivat@cisco.com>

   I'm fine with spinning it this way. It avoids using the *term* B2BUA, 
   but it still is talking about a B2BUA since it is talking about a UA 
   that has received a response from downstream of itself. So we are 
   playing political correctness gavmes. But I'm ok with that.

More importantly, the rule I suggest handles proxies, UAs, B2BUAs,
gateways, etc. *uniformly*.  (I use the term "agent" for "all devices
which send and receive SIP".)  There's no point attempting to make
general statements about the consequences of the rule for particular
classes of devices, as that only risks making a mistake by overlooking
unusual cases.  The rule states exactly when a 199 should be sent.

Dale
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We've been batting INFO around for many, many years now (like, 10 --  
it predates the SIP working group). A couple of meetings back, we  
agreed that we would discuss use cases for INFO packages, and if we  
didn't find any consensus there, then we would go ahead and publish an  
"INFO Considered Harmful" RFC.

We tried having this discussion at the last IETF, but that just didn't  
work out.

Do we try again, or just give up and publish "INFO Considered Harmful"?

Personally, I don't care anymore -- I just want to drive a stake into  
the heart of this undead-thing, cut off its head, stuff the mouth with  
holy wafers, and bury it at a crossroads somewhere.

But this WG only slays by consensus, so what do you want?

--
Dean

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Cc: sip@ietf.org
Subject: Re: [Sip] INFO and what to do about it?
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Thanks for raising this, Dean.

I think the issue here is not so much 'what do we do about INFO', but 
rather, what are the actual extensibility models people are using with 
SIP and are we properly providing the techniques to make it all 
interoperate. In my mind, that is a really important issue.

In terms of what to do, I think we should absolutely define an INFO 
framework along the lines of Hadriel's draft, and we should produce a 
guidelines document which helps people figure out whether to use INFO, 
event packages, or something along the lines of a media framework like I 
had proposed with TOTE (separate question on whether to pursue TOTE or 
not).

I think the huge number of actual INFO usages, along with the many 
reasonable use cases we have identified, make a clear statement that 
this is something that is needed.

-Jonathan R.

Dean Willis wrote:
> We've been batting INFO around for many, many years now (like, 10 --  
> it predates the SIP working group). A couple of meetings back, we  
> agreed that we would discuss use cases for INFO packages, and if we  
> didn't find any consensus there, then we would go ahead and publish an  
> "INFO Considered Harmful" RFC.
> 
> We tried having this discussion at the last IETF, but that just didn't  
> work out.
> 
> Do we try again, or just give up and publish "INFO Considered Harmful"?
> 
> Personally, I don't care anymore -- I just want to drive a stake into  
> the heart of this undead-thing, cut off its head, stuff the mouth with  
> holy wafers, and bury it at a crossroads somewhere.
> 
> But this WG only slays by consensus, so what do you want?
> 
> --
> Dean
> 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   499 Thornall St.
Cisco Fellow                                   Edison, NJ 08837
Cisco, Voice Technology Group
jdrosen@cisco.com
http://www.jdrosen.net                         PHONE: (408) 902-3084
http://www.cisco.com
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This list is for NEW development of the core SIP Protocol
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Cc: sip@ietf.org, Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] Toward the Evolution of SIP and Related Working Groups
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I agree with Mary - this seems like its just moving work around.

To reiterate the mantra of SIPPING itself - what PROBLEM are you trying 
to solve Dean? Is it that there is not enough people to do the work? As 
Mary points out, your suggestion only makes it worse. Is it to get more 
meeting slots? Is it to better align IETF work with the needs of the 
actual marketplace?

Thanks,
Jonathan R.

Mary Barnes wrote:
> I have a few comments below [MB].  And one more mega comment/proposal
> here.  Mobile IP is an area that has a similar history as SIP in terms
> of the demand and explosion of work. They considered our SIP/SIPPING
> model (at that time we had few other spinoffs, as I recall) when they
> split their WGs. They have a group for IPv6 work(mext), one for
> IPv4(mipv4) and one for performance, signaling and
> optimizations(mipshop). So, from you list of numbers, I could also see a
> 3 WG split for SIP/SIPPING (discounting all the current spinoffs) using
> your #s (and assuming SIP WG is done):
> A) #2+#3: SIP core protocol and extensions to fold into next protocol
> version
> B) #4:  Policies and security related aspects of SIP Protocol and then
> put the burden of screening new work such as SPIT, identity, etc. in
> this group until it's decided that the workload demands a separate WG.  
> C) #5: Basically, SIPPING slightly reduced from current charter  -
> filtering  (including OPS related items), howtos, discussing p-headers,
> etc. - the primary difference here is that this group would not screen
> basic SIP extension proposals or any of the policy/security, etc.
> related work. Right now we don't do the security, but some of the other
> stuff has been debatable.  
> 
> In my mind, the primary thing this would accomplish would be separating
> out the policy and security work to ensure there's the right focus and
> attention. I also think this makes scheduling only slightly easier in
> that it might be easier to ensure security folks could attend. 
> 
> However, I don't think this really helps that much with overall workload
> - it's just moving some things around. For example,  if A above does all
> the SIP extension screening, that workload may make A be just as crazy
> as SIP is now OR if in the end the security stuff is causing the
> craziness, then group B becomes the crazy one, until it sheds more work
> to new WGs and of course C stays fairly sane, which is what I think it
> is now ;)  
> 
> In general, I agree that SIP should follow the model that SIPPING did a
> couple years ago and NOT accept new work items until currently chartered
> items are done (or at least down to a manageable handful (5-6)) rather
> than the current 17 (use the tools page rather than WG charter to see
> the delta), so putting the WG on a path to closure is an excellent idea.
> 
> 
> Mary.
> 
> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Dean Willis
> Sent: Monday, June 16, 2008 6:23 PM
> To: sip@ietf.org
> Subject: [Sip] Toward the Evolution of SIP and Related Working Groups
> 
> 
> The SIP working group spun off from the MMUSIC working group when we
> realized that the SIP work was reasonably separable from the other work
> going on in MMUSIC.
> 
> The SIP working group first met officially at IETF 46 in November 1999,
> although we did have an earlier interim meeting.
> 
> This means that the fall 1999 meeting will be our 10th anniversary. In
> addition to having a big party, I'd also like to plan for this being the
> last meeting of the SIP working group as it is now defined. Ten years is
> long enough. It's time we restructured.
> 
> There will probably still be people who want to change, revise, extend,
> or otherwise manipulate the family of SIP specifications after 2009.
> 
> So what should we do? How should we organize to get those needs met?   
> Rest assured, we'll be having further discussions on this topic in the
> coming months. But I'd like to start the discussion now.
> 
> By November 2009, I fully expect SIP to have completed the majority of
> currently chartered items. Between now and then, we'll probably also
> take on and complete a bit of new work (although I plan to be very
> resistant to new work). I also expect that there are a few items on our
> charter now that we're not going to finish. It may well be that some of
> those items need a different organizational structure to progress than
> what we have. Or maybe they don't need doing.
> 
> Further, I've come to believe that most working groups should be
> shorter-lived and tightly constrained by their charters to produce a
> single concise set of deliverables. There's still a need for a longer-
> lived thing that maintains architectural oversight. I've come to believe
> that this is the "area", not the working group.
> 
> Let's start the recap by looking at the current charter of the SIP WG.  
> I've eliminated the WGLC milestones to shorten the list, as each has a
> following IESG milestone. Note that there's a Feb 2008 milestone for
> "Revise Charter". You can consider this topic the first installment on
> that milestone, even though it is a bit late.
> 
> 1) Jul 2007 Location Conveyance with SIP to IESG (PS)
> 
> 2) Aug 2007 Connection reuse mechanism to IESG (PS)
> 
> 3) Sep 2007 Session Policies to IESG as PS
> 
> 4) Sep 2007 Example security flows to IESG (Informational)
> 
> 5) Oct 2007 New resource priority namespaces for DISA to IESG (PS)
> 
> 6) Nov 2007 Requirements for media keying to IESG (Informational)
> 
> 7 )Dec 2007 Extensions to SIP UA Profile Delivery Change Notification
> Event Package for XCAP to IESG (PS)
> 
> 8) Dec 2007 Roadmap for SIP to IESG (Informational)
> 
> 9) Dec 2007 Using SAML for SIP to IESG (PS)
> 
> 10) Dec 2007 Extension for use in etags in conditional notification to
> IESG (PS)
> 
> 11) Feb 2008 Identify requirements for test matrix to move SIP to Draft
> Standard
> 
> 12) Feb 2008 Delivering request-URI and parameters to UAS via proxy to
> IESG (PS)
> 
> 13) Feb 2008 Revise charter
> 
> 14) Feb 2008 Establishment of secure media sessions using DTLS-SRTP to
> IESG (PS)
> 
> 15) Apr 2008 MIME body handling in SIP to IESG (PS)
> 
> 16) Apr 2008 Essential corrections to RFC 3261 (1st batch) to IESG (PS)
> 
> 17) Jul 2008 Mechanisms for UA initiated privacy to IESG (BCP)
> 
> 18) Aug 2008 Guidelines for double route recording to IESG (BCP)
> 
> 19) Sep 2008 X.509 Certificates for TLS use in SIP to IESG (PS)
> 
> 20) Sep 2008 X.509 extended key usage for SIP to IESG (PS)
> 
> 21) Nov 2008 Termination of early dialog prior to final response to IESG
> (PS)
> 
> 
> Looking at this, we have a couple of discrete tasks that I think might
> make the basis for different narrowly-scoped working groups. We've also
> got a lot of stuff that we're late on delivering!
> 
> 1) SIP WG (to finish by end of 2009) gets the bulk of these items.
> 
> 2) SIP Draft Standard WG: Takes the milestone of "Identify requirements
> for test matrix to move SIP to Draft Standard" and all of the follow-on
> work needed to move SIP from Proposed to Draft on the standards track.
> There's certainly enough work here for a dedicated working group, with
> the last milestone being something like "Deliver SIP as a Draft
> Standards Document to IESG". We may wish to consider whether we really
> want to go this route or whether we should instead be thinking of a "SIP
> 3.0" effort. My personal though is that there's just too much cruft in
> the RFC 3261 family to make Draft, but I could be proven wrong. I'd
> rather see us put SIP 2.0 into maintenance mode at the end of 2009 and
> move the bulk of our resources into developing a SIP 3.0 family designed
> from the ground up to be Draft-Standard achievable.
> [MB] I like this suggestion. [/MB]
> 
> 3) RAI Essential Corrections WG: Does this justify its own working
> group? Remember, it would be not just RFC 3261, but the entire family of
> related RFCs that would need lock-step maintenance and corrections.  
> Another approach might be to roll the corrections into the Draft
> Standard version of SIP 2.0.
> [MB] I think this should be roleed into 2) above. [/MB]
> 
> 
> 4) RAI Policies WG: We have these ongoing, not-quite-committed work
> items like session policies, use of SAML, etc.  that we've never been
> able to get fully wrapped up. I suspect they need their own WG. Or we
> need to decide that these things just aren't worth doing and stop
> feeling guilty about not finishing. As it is, I don't have a good
> feeling about SIP finishing them.
> [MB] I like this idea and would it also not cover any outstanding
> ongoing security items or are you anticipating that to addressed
> entirely by the currently chartered work items?  [/MB]
> 
> 5) SIP Operations: I'd like to see SIPPING replaced with a "how to"  
> group possibly under the OPS banner. They could deal with all the "how
> do we do X with SIP" from other SDOs. When confronted by a need to
> extend SIP to meet a given need, they could then BOF up a new WG to deal
> with the required extension.
> [MB] Are you proposing this to replace SIPPING WG?  If so, I don't see
> it falling under the banner of OPS. I do see that there may potentially
> be the need to initially discuss OPS related items, but as in the past,
> I see those being completed in the appropriate OPS WGs (e.g.,
> performance).  I have mixed views on how effective it would be to have a
> BoF and new WG for new extensions. I don't think the model we currently
> have is that ineffective in terms of first discussing requirements
> before agreeing a solution.  I see a SIP Maintenace group perhaps to
> handle these items, although in many cases, I think it would be far more
> effective to roll these things into your item 2 - i.e., don't you see
> SIP 3.0 as pulling in a lot of the headers that have been defined since
> RFC 3261 has been published. [/MB]
> 
> We already have working groups looking at conferencing, peering,
> telephony feature interop, provisioning, peer-to-peer, NAT traversal,
> 
> Other things that might need WGs: SPIT? Identity and Reputation?
> 
> [MB] I would agree these two topics might warrant separate WGs, although
> some of that work overlaps with your WG 4, I believe, the division of
> work may not be so clear.  [/MB]
> 
> I also suspect we need a couple of directorates that are not WGs to act
> in an advisory capacity:
> 
> [MB] I think we already have these and more covered by our current RAI
> area review team. These are specialized reviews and assignment of
> reviewers is based on area of expertise and we've got security and event
> package gurus already on the team. [/MB]
> 
> 1) RAI Security Directorate: Focuses on security review and
> architectural issues in RAI documents.
> 
> 2) RAI Event Package Directorate: Reviews and approves new SIP event
> packages
> 
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use
> sip-implementors@cs.columbia.edu for questions on current sip Use
> sipping@ietf.org for new developments on the application of sip
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   499 Thornall St.
Cisco Fellow                                   Edison, NJ 08837
Cisco, Voice Technology Group
jdrosen@cisco.com
http://www.jdrosen.net                         PHONE: (408) 902-3084
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: Re: [Sip] INFO and what to do about it?
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Dean Willis wrote:
> We've been batting INFO around for many, many years now (like, 10 --  
> it predates the SIP working group). A couple of meetings back, we  
> agreed that we would discuss use cases for INFO packages, and if we  
> didn't find any consensus there, then we would go ahead and publish an  
> "INFO Considered Harmful" RFC.
> 
> We tried having this discussion at the last IETF, but that just didn't  
> work out.
> 
> Do we try again, or just give up and publish "INFO Considered Harmful"?
> 
> Personally, I don't care anymore -- I just want to drive a stake into  
> the heart of this undead-thing, cut off its head, stuff the mouth with  
> holy wafers, and bury it at a crossroads somewhere.

I gave my try, very unsuccessfully. So I just want to remind you not to 
forget the garlic, the silver cross, and the sunlight.

> But this WG only slays by consensus, so what do you want?
> 
> --
> Dean
> 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
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Some comments on this draft. I apologize if they are duplicates.

Section 5 should only allow the UAS to send a 199 if it sent a dialog creating (101-199) provisional response. I know its implicit, but needs to be explicitly stated to prevent extra traffic. Suggest something like this:

   When a UAS wants to terminate an INVITE server transaction it sends a non-2xx
   SIP final response, as specified in [RFC3261].  In addition, if the UAS has
   previously sent a dialog creating 1xx response (101-199) for the transaction,
   then prior to sending the non-2xx SIP final response, the UAS MAY send a 199
   response to indicate that the early dialog is being terminated. The 199 response
   MUST have the same tag in the To header as sent in the dialog creating 1xx
   response.

In section 6 requires a forking proxy to send 199s for all early dialogs associated with the non-2xx response. However, if there are multiple forking proxies along the path (or some UASs send 199s), that could result in multiple 199's for a given dialog. IMO, if the proxy has already forwarded a 199 for a given dialog, it should not send one of its own.

On the topic of UAS's sending 199, I actually think things will work better if they did. I know there has been lots of discussion about B2BUAs doing it, but consider a UAS that is aware that it has multiple registered contacts (via presence or registration event package or other means), it might want to send the 199 to be sure that it does reach the UAC (in case the proxy does not support it).

Section 8 Open Issues:

IMO, the 199's To-tag should identify the dialog being terminated.

IMO, the 199's body MAY contain message/sipfrag with the non-2xx response code and important headers like Warning.

cheers,
(-:bob

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Internet-Drafts@ietf.org
Sent: Friday, May 23, 2008 1:15 PM
To: i-d-announce@ietf.org
Cc: sip@ietf.org
Subject: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt

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

        Title           : Response Code for Indication of Terminated Dialog
        Author(s)       : C. Holmberg
        Filename        : draft-ietf-sip-199-00.txt
        Pages           : 9
        Date            : 2008-5-15

This specification defines a new SIP response code, 199 Early Dialog
   Terminated, which a SIP entity can use to indicate upstream that an
   early dialog has been terminated.  The response code can be used by a
   SIP entity, normally a forking SIP proxy, to allow the UAC to know
   that an early dialog has been terminated before it a final response
   is sent to the request.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-199-00.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

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Internet-Draft.
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SIP is actually an excellent success story. Not only is SIP the world
standard but there are countless SIP implementations as OS SW.

Maybe we could agree on the following to declare victory and preserve its
success:

1. Any SIP and related work in the IETF is focused on the (stupid = no
intelligence) Internet, as defined in AIB issued RFCs, keeping the e2e
principle in mind.

2. Service provider specific protocols (to support "network based services")
belong in ITU-T or other similar service provider focused organizations,

3. The main deliverable of a WG is free software to validate and support the
protocol document and its refinement. If no free software is contributed,
the WG activity should be suspended. Internet-Drafts not supported by freely
available software (that can be inspected and tried out) and measurements
cannot be proven right or wrong and should therefore be ignored.
There is plenty of IEEE and University work to prove this approach, which
was also the basis for the rise and success of the IETF.

Thanks (and feel free to flame),

Henry 


On 6/16/08 6:23 PM, "Dean Willis" <dean.willis@softarmor.com> wrote:

> 
> The SIP working group spun off from the MMUSIC working group when we
> realized that the SIP work was reasonably separable from the other
> work going on in MMUSIC.
> 
> The SIP working group first met officially at IETF 46 in November
> 1999, although we did have an earlier interim meeting.
> 
> This means that the fall 1999 meeting will be our 10th anniversary. In
> addition to having a big party, I'd also like to plan for this being
> the last meeting of the SIP working group as it is now defined. Ten
> years is long enough. It's time we restructured.
> 
> There will probably still be people who want to change, revise,
> extend, or otherwise manipulate the family of SIP specifications after
> 2009.
> 
> So what should we do? How should we organize to get those needs met?
> Rest assured, we'll be having further discussions on this topic in the
> coming months. But I'd like to start the discussion now.
> 
> By November 2009, I fully expect SIP to have completed the majority of
> currently chartered items. Between now and then, we'll probably also
> take on and complete a bit of new work (although I plan to be very
> resistant to new work). I also expect that there are a few items on
> our charter now that we're not going to finish. It may well be that
> some of those items need a different organizational structure to
> progress than what we have. Or maybe they don't need doing.
> 
> Further, I've come to believe that most working groups should be
> shorter-lived and tightly constrained by their charters to produce a
> single concise set of deliverables. There's still a need for a longer-
> lived thing that maintains architectural oversight. I've come to
> believe that this is the "area", not the working group.
> 
> Let's start the recap by looking at the current charter of the SIP WG.
> I've eliminated the WGLC milestones to shorten the list, as each has a
> following IESG milestone. Note that there's a Feb 2008 milestone for
> "Revise Charter". You can consider this topic the first installment on
> that milestone, even though it is a bit late.
> 
> 1) Jul 2007 Location Conveyance with SIP to IESG (PS)
> 
> 2) Aug 2007 Connection reuse mechanism to IESG (PS)
> 
> 3) Sep 2007 Session Policies to IESG as PS
> 
> 4) Sep 2007 Example security flows to IESG (Informational)
> 
> 5) Oct 2007 New resource priority namespaces for DISA to IESG (PS)
> 
> 6) Nov 2007 Requirements for media keying to IESG (Informational)
> 
> 7 )Dec 2007 Extensions to SIP UA Profile Delivery Change Notification
> Event Package for XCAP to IESG (PS)
> 
> 8) Dec 2007 Roadmap for SIP to IESG (Informational)
> 
> 9) Dec 2007 Using SAML for SIP to IESG (PS)
> 
> 10) Dec 2007 Extension for use in etags in conditional notification to
> IESG (PS)
> 
> 11) Feb 2008 Identify requirements for test matrix to move SIP to
> Draft Standard
> 
> 12) Feb 2008 Delivering request-URI and parameters to UAS via proxy to
> IESG (PS)
> 
> 13) Feb 2008 Revise charter
> 
> 14) Feb 2008 Establishment of secure media sessions using DTLS-SRTP to
> IESG (PS)
> 
> 15) Apr 2008 MIME body handling in SIP to IESG (PS)
> 
> 16) Apr 2008 Essential corrections to RFC 3261 (1st batch) to IESG (PS)
> 
> 17) Jul 2008 Mechanisms for UA initiated privacy to IESG (BCP)
> 
> 18) Aug 2008 Guidelines for double route recording to IESG (BCP)
> 
> 19) Sep 2008 X.509 Certificates for TLS use in SIP to IESG (PS)
> 
> 20) Sep 2008 X.509 extended key usage for SIP to IESG (PS)
> 
> 21) Nov 2008 Termination of early dialog prior to final response to
> IESG (PS)
> 
> 
> Looking at this, we have a couple of discrete tasks that I think might
> make the basis for different narrowly-scoped working groups. We've
> also got a lot of stuff that we're late on delivering!
> 
> 1) SIP WG (to finish by end of 2009) gets the bulk of these items.
> 
> 2) SIP Draft Standard WG: Takes the milestone of "Identify
> requirements for test matrix to move SIP to Draft Standard" and all of
> the follow-on work needed to move SIP from Proposed to Draft on the
> standards track. There's certainly enough work here for a dedicated
> working group, with the last milestone being something like "Deliver
> SIP as a Draft Standards Document to IESG". We may wish to consider
> whether we really want to go this route or whether we should instead
> be thinking of a "SIP 3.0" effort. My personal though is that there's
> just too much cruft in the RFC 3261 family to make Draft, but I could
> be proven wrong. I'd rather see us put SIP 2.0 into maintenance mode
> at the end of 2009 and move the bulk of our resources into developing
> a SIP 3.0 family designed from the ground up to be Draft-Standard
> achievable.
> 
> 3) RAI Essential Corrections WG: Does this justify its own working
> group? Remember, it would be not just RFC 3261, but the entire family
> of related RFCs that would need lock-step maintenance and corrections.
> Another approach might be to roll the corrections into the Draft
> Standard version of SIP 2.0.
> 
> 4) RAI Policies WG: We have these ongoing, not-quite-committed work
> items like session policies, use of SAML, etc.  that we've never been
> able to get fully wrapped up. I suspect they need their own WG. Or we
> need to decide that these things just aren't worth doing and stop
> feeling guilty about not finishing. As it is, I don't have a good
> feeling about SIP finishing them.
> 
> 5) SIP Operations: I'd like to see SIPPING replaced with a "how to"
> group possibly under the OPS banner. They could deal with all the "how
> do we do X with SIP" from other SDOs. When confronted by a need to
> extend SIP to meet a given need, they could then BOF up a new WG to
> deal with the required extension.
> 
> We already have working groups looking at conferencing, peering,
> telephony feature interop, provisioning, peer-to-peer, NAT traversal,
> 
> Other things that might need WGs: SPIT? Identity and Reputation?
> 
> I also suspect we need a couple of directorates that are not WGs to
> act in an advisory capacity:
> 
> 1) RAI Security Directorate: Focuses on security review and
> architectural issues in RAI documents.
> 
> 2) RAI Event Package Directorate: Reviews and approves new SIP event
> packages
> 
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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Hi,

I agree with Jonathan.

Regards,

Christer =


-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Jonat=
han Rosenberg
Sent: 18. kes=E4kuuta 2008 1:32
To: Dean Willis
Cc: sip@ietf.org
Subject: Re: [Sip] INFO and what to do about it?

Thanks for raising this, Dean.

I think the issue here is not so much 'what do we do about INFO', but rathe=
r, what are the actual extensibility models people are using with SIP and a=
re we properly providing the techniques to make it all interoperate. In my =
mind, that is a really important issue.

In terms of what to do, I think we should absolutely define an INFO framewo=
rk along the lines of Hadriel's draft, and we should produce a guidelines d=
ocument which helps people figure out whether to use INFO, event packages, =
or something along the lines of a media framework like I had proposed with =
TOTE (separate question on whether to pursue TOTE or not).

I think the huge number of actual INFO usages, along with the many reasonab=
le use cases we have identified, make a clear statement that this is someth=
ing that is needed.

-Jonathan R.

Dean Willis wrote:
> We've been batting INFO around for many, many years now (like, 10 -- =

> it predates the SIP working group). A couple of meetings back, we =

> agreed that we would discuss use cases for INFO packages, and if we =

> didn't find any consensus there, then we would go ahead and publish an =

> "INFO Considered Harmful" RFC.
> =

> We tried having this discussion at the last IETF, but that just didn't =

> work out.
> =

> Do we try again, or just give up and publish "INFO Considered Harmful"?
> =

> Personally, I don't care anymore -- I just want to drive a stake into =

> the heart of this undead-thing, cut off its head, stuff the mouth with =

> holy wafers, and bury it at a crossroads somewhere.
> =

> But this WG only slays by consensus, so what do you want?
> =

> --
> Dean
> =

> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use =

> sip-implementors@cs.columbia.edu for questions on current sip Use =

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


-- =

Jonathan D. Rosenberg, Ph.D.                   499 Thornall St.
Cisco Fellow                                   Edison, NJ 08837
Cisco, Voice Technology Group
jdrosen@cisco.com
http://www.jdrosen.net                         PHONE: (408) 902-3084
http://www.cisco.com
_______________________________________________
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This list is for NEW development of the core SIP Protocol Use sip-implement=
ors@cs.columbia.edu for questions on current sip Use sipping@ietf.org for n=
ew developments on the application of sip
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Bob Penfield wrote:

> On the topic of UAS's sending 199, I actually think things will work better if they did. I know there has been lots of discussion about B2BUAs doing it, but consider a UAS that is aware that it has multiple registered contacts (via presence or registration event package or other means), it might want to send the 199 to be sure that it does reach the UAC (in case the proxy does not support it).

Bob, I don't understand the point you are making. Can you explain further?

Are you talking about a case where there is a forking proxy in front of 
the UAS, and the proxy doesn't know about 199, and the UAS wants to send 
the 199 to give the UAC a warning in case the final response is delayed 
by the proxy?

While there may be some cases where that would be advantageous, in many 
others it would just increase the message traffic for every failing 
call. And I don't see how the UAS would be able to discern the cases 
where it would be advantageous. I guess it could be configured in those 
cases where all inbound calls are forked, but how many such cases are there?

	Paul
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On Jun 17, 2008, at 5:37 PM, Jonathan Rosenberg wrote:

> I agree with Mary - this seems like its just moving work around.
>
> To reiterate the mantra of SIPPING itself - what PROBLEM are you  
> trying to solve Dean? Is it that there is not enough people to do  
> the work? As Mary points out, your suggestion only makes it worse.  
> Is it to get more meeting slots? Is it to better align IETF work  
> with the needs of the actual marketplace?


The problem that I'm trying to solve is making the task space  
manageable.

The first thing is to distribute the chair load. Everybody says "The  
problem isn't the chairs, it;s the lack of reviewers". Well, everybody  
but the chairs say that, actually. If you look at the current status  
from the beginning of this thread, you'll see that Keith and I are the  
long poles in the SIP tent.  Sometimes stuff waits a year for us to  
finish the proto writeup -- usually because when we get to doing the  
writeup, we find a bajillion errors the reviewers and authors missed.  
I typically see drafts revved at least twice during the writeup cycle,  
and that's a heck of a lot of work on the part of the chairs!

The second problem is "learning to say no". I think WGs should have  
very narrowly defined charters and that it should be very, very  
difficult to add substantial new work to a charter. That is, it should  
be nearly as difficult to add work to a WG as it is to form up a new  
WG, and that often, the correct answer actually is "form a new WG".  
Perhaps even more often, the correct answer is "no".

I think developing a SIP extension should be a Big Thing. If it's  
important enough to develop a SIP extension for, it is probably  
important enough to charter a working group to develop that extension.  
It's a Really Bad Idea to have a standing working group that sits  
around and develops SIP extensions. All that gets us is a whole lot of  
extensions. It doesn't get us any closer to a draft standard.

Working groups shouldn't last ten years. Only in rare circumstances  
should they last two years, and the average should be more like one  
year.

The key result of this sort of reorg is that a WG, and most especially  
the chairs of that WG, can be narrowly focused on a small and  
manageable set of deliverables. We should give them the luxury of  
doing one job and doing it well. Chairs should be able to do a real  
day-to-day job and run a working group on the side. I can tell you,  
SIP doesn't work that way now! Even the lousy job I do requires at  
least 50% of my working time, and that's going to break the bank  
eventually.

--
Dean

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Subject: [Sip] Cut-off dates reminder for IETF 72
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Mary said this beautifully on SIPPING, so I'll just borrow her text  
and repost it here:

This is a reminder of the cut-off dates for draft submission for  
IETF-72:

- Monday, July 7th for -00s

- Monday, July 14th for -01+

Refer here for complete details:
http://www.ietf.org/meetings/72/72-cutoff_dates.html

So, these dates are just a few weeks off and our WG agenda is due on
July 16th (less than a month away).

Also, please use the tool to upload your document:

https://datatracker.ietf.org/idst/upload.cgi

It will be published very, very quickly this way, but I will note that  
it does not check for ALL the nits, so it's a good idea to run idnits
before submitting:

http://tools.ietf.org/tools/idnits/


--
Dean

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From: Bob Penfield <BPenfield@acmepacket.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
Date: Wed, 18 Jun 2008 08:03:33 -0400
Thread-Topic: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt
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> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Tuesday, June 17, 2008 11:44 PM
> To: Bob Penfield
> Cc: sip@ietf.org
> Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt
>
>
> Bob Penfield wrote:
>
>> On the topic of UAS's sending 199, I actually think things will work
>> better if they did. I know there has been lots of discussion about
>> B2BUAs doing it, but consider a UAS that is aware that it has multiple
>> registered contacts (via presence or registration event package or
>> other means), it might want to send the 199 to be sure that it does
>> reach the UAC (in case the proxy does not support it).
>
> Bob, I don't understand the point you are making. Can you explain further?
>
> Are you talking about a case where there is a forking proxy in front of
> the UAS, and the proxy doesn't know about 199, and the UAS wants to send
> the 199 to give the UAC a warning in case the final response is delayed
> by the proxy?

Yes, that is pretty much the case I was considering.

>
> While there may be some cases where that would be advantageous, in many
> others it would just increase the message traffic for every failing
> call. And I don't see how the UAS would be able to discern the cases
> where it would be advantageous. I guess it could be configured in those
> cases where all inbound calls are forked, but how many such cases are there?
>

If the UAS knows that the AOR has multiple registered contacts, it knows that inbound calls will be forked. In that case, it could send the 199. As long as proxies don't generate 199s in addition to ones they forward, the increased traffic could be limited.

Having proxies generate responses on behalf of a UAS is a bit of a violation of the end-to-end principle from a purist point of view. By certainly having only proxies generate 199s would limit the increase in messages.

I was just trying to point out that there might be cases where real User Agent would send 199 and not just B2BUAs.


>        Paul
>
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Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt
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Bob Penfield wrote:
> 
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>> Sent: Tuesday, June 17, 2008 11:44 PM
>> To: Bob Penfield
>> Cc: sip@ietf.org
>> Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-199-00.txt
>>
>>
>> Bob Penfield wrote:
>>
>>> On the topic of UAS's sending 199, I actually think things will work
>>> better if they did. I know there has been lots of discussion about
>>> B2BUAs doing it, but consider a UAS that is aware that it has multiple
>>> registered contacts (via presence or registration event package or
>>> other means), it might want to send the 199 to be sure that it does
>>> reach the UAC (in case the proxy does not support it).
>> Bob, I don't understand the point you are making. Can you explain further?
>>
>> Are you talking about a case where there is a forking proxy in front of
>> the UAS, and the proxy doesn't know about 199, and the UAS wants to send
>> the 199 to give the UAC a warning in case the final response is delayed
>> by the proxy?
> 
> Yes, that is pretty much the case I was considering.
> 
>> While there may be some cases where that would be advantageous, in many
>> others it would just increase the message traffic for every failing
>> call. And I don't see how the UAS would be able to discern the cases
>> where it would be advantageous. I guess it could be configured in those
>> cases where all inbound calls are forked, but how many such cases are there?
>>
> 
> If the UAS knows that the AOR has multiple registered contacts, it knows that inbound calls will be forked. In that case, it could send the 199. As long as proxies don't generate 199s in addition to ones they forward, the increased traffic could be limited.

How often does a UAS know that there are multiple contacts for this AOR?

Even if it does, it presumably can't know if it is the last one tried, 
the first, or whatever.

> Having proxies generate responses on behalf of a UAS is a bit of a violation of the end-to-end principle from a purist point of view. By certainly having only proxies generate 199s would limit the increase in messages.

Yes, it is.

> I was just trying to point out that there might be cases where real User Agent would send 199 and not just B2BUAs.

I do appreciate the point that there *can* be cases where the UAS would 
have enough info to justify sending a 199 response. But the cases where 
a UAS is likely to know that in practice seem limited, with the 
exception of B2BUAs. I would just like to avoid giving UA developers the 
idea that they *ought* to be generating 199 responses on a regular 
basis. Doing so "to be helpful" is likely to be sub-optimal in many cases.

>>        Paul
>>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
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From sip-bounces@ietf.org  Wed Jun 18 12:42:56 2008
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From: "Mary Barnes" <mary.barnes@nortel.com>
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Subject: Re: [Sip] INFO and what to do about it?
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As Jonathan noted, there is a clear need for something, as I don't see
folks giving up on using INFO at all and providing some
structure/standardization around the use is a good thing. The following
documents are a good starting place IMHO:
http://tools.ietf.org/id/draft-kaplan-sip-info-use-cases-01.txt
http://tools.ietf.org/id/draft-kaplan-sip-info-events-01.txt
http://tools.ietf.org/id/draft-kaplan-sipping-dtmf-package-00.txt

As far as providing guidelines on what approaches folks should use
depending upon requirements and problem statements, I agree with
Jonathan that a document of this sort would be useful, perhaps using
this document as a starting point:
http://tools.ietf.org/id/draft-rosenberg-sip-info-litmus-00.txt 

We would of course need to decide how to partition the work either using
the traditional SIP/SIPPING split of new SIP headers, etc. in SIP and
event packages, informational docs, etc. in SIPPING or possibly consider
this as a test case for your (Dean's) proposal of starting new WGs for
new work items, since it would seem that in the end there could be a
reasonable handful of work items to progress and the work could be
fairly tightly scoped. Although, I fear this group would have the same
difficulty in reaching a stopping point as SIP/SIPPING in terms of folks
wanting to define new INFO event packages.

Likely, the most contentious topic of debate is whether we need to
define a general extensibility model.  Given we don't have a lot of use
cases (other than cases where we've already completed work or well on
the way to completing), it's kinda tough to say that if we build it,
folks will use it.  So, personally, my opinion would be to defer this
topic until we see a real need, although I realize there's a risk of not
being able to deliver this in a timely manner if we wait, particularly
given past history on how quickly we can complete work items.  However,
in one sense I feel that it's just way too late to do this sort of thing
(at least in SIP 2.0). 

Mary. 

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Dean Willis
Sent: Tuesday, June 17, 2008 5:27 PM
To: sip@ietf.org
Subject: [Sip] INFO and what to do about it?


We've been batting INFO around for many, many years now (like, 10 -- it
predates the SIP working group). A couple of meetings back, we agreed
that we would discuss use cases for INFO packages, and if we didn't find
any consensus there, then we would go ahead and publish an "INFO
Considered Harmful" RFC.

We tried having this discussion at the last IETF, but that just didn't
work out.

Do we try again, or just give up and publish "INFO Considered Harmful"?

Personally, I don't care anymore -- I just want to drive a stake into
the heart of this undead-thing, cut off its head, stuff the mouth with
holy wafers, and bury it at a crossroads somewhere.

But this WG only slays by consensus, so what do you want?

--
Dean

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   From: Henry Sinnreich <hsinnrei@adobe.com>

   3. The main deliverable of a WG is free software to validate and support the
   protocol document and its refinement.

A good concept, but I would amend this to "... two interoperable and
idependenly-implemented free softwares [grammar?] ...", as it was in
the Good Old Days.

Dale
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Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
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   From: "Christer Holmberg" <christer.holmberg@ericsson.com>

Regarding the few open items:

   >* Section 1
   >
   >   When a forking proxy receives a non-2xx final response, it does not
   >   always immediately forward the response upstream towards the sender
   >   of the associated request.  Instead, the forking proxy "stores" it
   >   and waits for further final responses from remote destinations where
   >   the forked request was forwarded.
   >
   >The text does not allow for serial forking, only parallel forking,
   which might be confusing to people who are more aware >of serial
   forking.  So amend the last sentence to:
   >
   >   Instead, the forking proxy "stores" it and waits for further final
   >   responses from remote destinations where the forked request was
   >   forwarded, and possibly forwards the request to additional
   >   destinations.

   [CHH] I can change the text. My intention was not to prevent serial
   forking. Because, no matter what type of forking it is, the forking
   proxy will "store" final responses from the different forked legs.

Yes, I am sure that you did not intend to prevent or ignore serial
forking.  But I am afraid that the original text might lead a naive
reader to assume that only parallel forking is relevant regarding 199.
Fortunately, we only need to add a few words to ensure that no such
misunderstanding happens.

   >* Section 3.
   >
   >   REQ 1: It must be possible to indicate to the UAC that an early
   >   dialog has been terminated before a final response is sent.
   >
   >Oddly, though everyone thinks they understand this sentence, it doesn't
   really say what we want.  To really ensure that 
   >the UAC understands the early dialog has been terminated before the UAS
   sends a final response, the UAS would have to 
   >send 199 *and get the PRACK saying that it was received* before sending
   a final response.  This may well be worse than 
   >the present state of affairs.
   >
   >What we really mean is:
   >
   >   REQ 1: It must be possible for a UAS or proxy to send an indication
   >   to the UAC that an early dialog has been terminated, which
   >   indication will not be delayed waiting for a final response from
   >   any UAS.

   [CHH] I hear what you are saying, but it is a copy of the requirement
   text we agreed to in SIPPING.

The odd thing is that although the mechanism described in the I-D
solves the problem we want to solve, the requirement *as stated* does
not describe the problem we want to solve, nor does the I-D satisfy
the requirement as stated.  It is easy, when reading the requirement
to mis-interpret it (I did so myself several times), but what it
says is not what we intend.  How do we proceed if what SIPPING
approved is not what we are interested in?

   >* Section 4
   >
   >Add this paragraph before the last paragraph:
   >
   >   A UAC that receives a 199 response for an early dialog MUST NOT
   >   send any further requests on that dialog, except for a PRACK if the
   >   199 response was sent reliably, and PRACKs for other reliable
   >   provisional responses that are to be acknowledged.
   >
   >This is copied from section 7 (Backward Compatibility), but since it is
   a constraint on UACs, it should also be stated in 
   >this section.

   [CHH] Whether the text should be in the document at all depends on if we
   allow 199 to be sent reliably in the first place. Based on the comments
   received so far we should not mandate 199 to be sent reliably, even if
   100rel is required by the UAC. But, the question if whether we want to
   FORBID sending it reliably.

That is true, if we forbid reliable 199, then this proposed paragraph
would be modified in parallel with the paragraph in section 7.

   >* Section 4
   >
   >   The UAC MAY use additional information (for example the final
   >   response which triggered the 199 response) received in the 199
   >   response when initiating new sessions, if it is possible to avoid
   the
   >   new session initiation request to be rejected.
   >
   >What is the meaning of the final clause?

   [CHH] Others have asked the same, so I guess it needs some modification.
   The meaning is that the information received in the 199 may be used to
   solve the HERFP problem. It is not in the scope of the document to solve
   that problem, just to mention that the information may be used for a
   future mechanism solving it.

How about:  "The UAC MAY use additional information (for example the
final response which triggered the 199 response) received in the 199
response for additional purposes."

   >* Section 5
   >
   >Add "early" as marked:
   >
   >  When a UAS wants to terminate *an early* dialog it sends a non-2xx
   >   SIP final response, as specified in [RFC3261].  In addition, prior
   >  to sending the non-2xx SIP final response, the UAS MAY send a 199
   >   response to indicate that the dialog is being terminated.
   >
   >This clarifies that 199 cannot be used to terminate a confirmed dialog.

   [CHH] I have nothing against your proposal as such, but is this marking
   method that you are proposing something we normally use in IETF specs?

I am using *...* to make clear the words I inserted.  I do not intend
that the "*" would remain in the text of the I-D.

   >(The reason a UAS may want to send a 199 is to provide this information
   even when the upstream proxy does not support 
   >199.)
   >
   >There is a subtlty to the second sentence:
   >
   >Sending the 199 *before* the final response is not necessary due to the
   semantics of 199 (as it will probably reach the 
   >UAC first anyway, and it would not matter if it did not), but rather to
   prevent the upstream proxy from generating a 199 
   >based on the final response, and then to transmit the UAS's 199
   immediately after -- if the UAS sends the 199 first, any 
   >upstream proxy will (most likely) see the 199 before seeing the final
   response (which might induce the proxy to send its 
   >own 199).

   [CHH] This depends on whether we should keep the UAS text in the first
   place, or only describe when proxies send 199. Based on the comments
   I've received so far, we should not describe UAS behavior for sending
   199.

My preference is to state the explicit rule for when to send 199 that
we've been discussing on the list.  That rule implies all of the
consequences we've discussed.  It also correctly handles various
peculiar situations whose proper handling might not be clear at first
analysis.

   Example:

   Suppose that before ringing, a UA plays a message to the caller
   through a separate early dialog.  So it sends the following messages:

       183 Session Progress, to-tag = A, with SDP A
       RTP, according to SDP A
       199 Early Dialog Terminated, to-tag = A
       180 Ringing, to-tag = B
       200 OK, to-tag = B, with SDP B

   Following the rule, it's clear that sending the 199 is acceptable (and
   it is useful).

   Of course, we could also analyze the UA as a proxy that forks the
   call serially to two UAs, one of which plays the announcement, and
   we would obtain the same result.  But it's easier to apply the rule
   to the UA as a whole.

   >* Section 8
   >
   >Clarify this sentence to:
   >
   >   The 199 Early Dialog Terminated response code allows a SIP entity
   >   to indicate to upstream entities that a specific early dialog has
   >   been terminated, before a final response reaches the upstream
   >   entities.
   >
   >Add the following:
   >
   >   The early dialog that has been terminated is identified by the
   Call-Id
   >   header and the to-tag value of the 199 response.
   >
   >   Future standards may define optional additional information carried
   >   in the headers and/or body of the response.

   [CHH] I am ok with the first sentence, but I don't understand the
   meaning of the other ("Future standards...") in this context.

It is to allow future standardization of additional headers or
body-content (e.g., to provide HERFP information).

   >* Section 9
   >
   >   A 199 Early Dialog Terminated provisional response MUST NOT contain
   a
   >   new SDP offer/answer message body, but the sender of the response
   MAY
   >   insert a copy of a previously sent offer/answer message body as
   >   otherwise allowed by the offer/answer rules for a provisional
   >   response.
   >
   >This rule seems excessively strict, as it seems OK to send a new SDP
   answer in a 199, even if the dialog will be 
   >immediately terminated.
   >Certainly, the UAC cannot tell the difference, as there could have been
   a previous but lost 1xx response carrying the 
   >same SDP.
   >
   >The interaction of 199 and SDP probably needs further study...

   [CHH] If we are not going to specify UAS sending of 199 then there will
   be no SDP in it. I don't think the forking proxy should store received
   SDPs, and then insert one when triggering the 199.

Originally, I saw no use for having an SDP body in a 199, but I saw no
reason to forbid it, either.

Except that I have changed my mind -- since we are reserving the body
of 199 for future standardization (especially, to carry the
corresponding failure response), we should avoid letting it be used
for other purposes.

Dale
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Subject: Re: [Sip] INFO and what to do about it?
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I asked:


> We've been batting INFO around for many, many years now (like, 10 --
> it predates the SIP working group). A couple of meetings back, we
> agreed that we would discuss use cases for INFO packages, and if we
> didn't find any consensus there, then we would go ahead and publish an
> "INFO Considered Harmful" RFC.
>
> We tried having this discussion at the last IETF, but that just didn't
> work out.
>
> Do we try again, or just give up and publish "INFO Considered  
> Harmful"?
>
> Personally, I don't care anymore -- I just want to drive a stake into
> the heart of this undead-thing, cut off its head, stuff the mouth with
> holy wafers, and bury it at a crossroads somewhere.
>
> But this WG only slays by consensus, so what do you want?

Several people (Jonathan, Mary, Christer, Paul) have responded to the  
list  with various things that added up to "Publish INFO packages AND  
document the extension models for SIP". So far, nobody has suggested  
otherwise.

Does anybody want to do anything else? Speak now or forever visualize  
whirled peas.

--
Dean


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Hi,

I have a question regarding the REFER method: is it a target refresh
method?

RFC 3515 doesn't say anything about it. However, the Contact header
seems to be MANDATED in the REFER request (eventhough not shown in the
examples in the draft).

And, chapter 4 of RFC 4488 says:

"This document defines a new SIP header field: "Refer-Sub". This header
field is meaningful and MAY be used with a REFER request and the
corresponding 2XX response only. This header field set to "false"
specifies that a REFER-Issuer requests that the REFER- Recipient doesn't
establish an implicit subscription and the resultant dialog. Note that
when using this extension, the REFER remains a target refresh request
(as in the default case -- when the extension is not used)."

So, I guess that means that REFER IS a target refresh method?

Regards,

Christer



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<TITLE>REFER a target-refresh</TITLE>
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<P><FONT SIZE=3D2 FACE=3D"Courier New">Hi,</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">I have a question regarding the =
REFER method: is it a target refresh method?</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">RFC 3515 doesn't say anything =
about it. However, the Contact header seems to be MANDATED in the REFER =
request (eventhough not shown in the examples in the draft).</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">And, chapter 4 of RFC 4488 =
says:</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">&quot;This document defines a new =
SIP header field: &quot;Refer-Sub&quot;. This header field is meaningful =
and MAY be used with a REFER request and the corresponding 2XX response =
only. This header field set to &quot;false&quot; specifies that a =
REFER-Issuer requests that the REFER- Recipient doesn't establish an =
implicit subscription and the resultant dialog. Note that when using =
this extension, the REFER remains a target refresh request (as in the =
default case -- when the extension is not used).&quot;</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">So, I guess that means that REFER =
IS a target refresh method?</FONT>
</P>

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

<P><FONT SIZE=3D2 FACE=3D"Courier New">Christer</FONT>
</P>
<BR>

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_______________________________________________
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Use sipping@ietf.org for new developments on the application of sip
--===============1210867394==--


From sip-bounces@ietf.org  Thu Jun 19 02:29:48 2008
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Thread-Topic: Progress draft-holmberg-sip-keep
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From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
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Cc: Christer Holmberg <christer.holmberg@ericsson.com>
Subject: [Sip] Progress draft-holmberg-sip-keep
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(As SIP WG cochair)

We have been asked by the author of 

http://www.ietf.org/internet-drafts/draft-holmberg-sip-keep-01.txt

Whether the SIP WG can progress this document.

Because this draft arose as a result of the discussion of outbound, and
indeed seems to reuse the requirements from outbound, and these
requirements never really got handled in the SIPPING WG, it has been
agreed with the SIPPING chairs that we will handle this entirely within
SIP.

Now in order to ask for charter milestones, and indeed when we finally
present this to IESG, we will be asked for the level of support in the
WG, which is also predicated on does this fix a real problem, or is it
just a corner case with limited application. So:

QUESTION 1 TO SIP WG: Are the use cases sufficiently important to
proceed with this draft? The document states:

   Chapter 3.5 of draft-ietf-sip-outbound-13 [I-D.ietf-sip-outbound]
   defines two keep-alive techniques.  Even though the keep-alive
   techniques are separated from the Outbound mechanism
   [I-D.ietf-sip-outbound], it is currently not possible to indicate
   support of the keep-alive techniques without also indicating support
   for the Outbound mechanism.

   The Outbound mechanism is enabled during the UA registration phase.
   However, there are use-cases where the UA does not register itself,
   but still needs to be able to make calls and maintain NAT bindings
   open during the duration of that call.  A typical example is
   emergency calls.  There are also cases where entities do not support
   the Outbound mechanism, but still want to be able to indicate support
   and use the keep-alive techniques defined in [I-D.ietf-sip-outbound].

At first sight this is not the most inspiring declaration of the need
for the document. Please respond indicating whether you consider this a
useful draft, and propose text that you think would be useful in this
section. Conversely, if you think this draft is not useful and the WG
has other more important things to work on first, please also respond.

QUESTION 2 TO SIP WG: Do we have a robust set of requirements for
proceeding with this work? The document currently lists:

   REQ 1: It MUST be possible for a UA to indicate support of the keep-
   alive techniques defined [I-D.ietf-sip-outbound] if the UA supports
   only the keep-alive part of [I-D.ietf-sip-outbound].

   REQ 2: It MUST be possible for an edge proxy to indicate support of
   the keep-alive techniques defined [I-D.ietf-sip-outbound] if the edge
   poxy supports only the keep-alive part of [I-D.ietf-sip-outbound].

It would be desirable to agree these at the outset, and not revisit them
if we continue with the work. So if you require clarification,
modification, or addition to these two requirements, then please also
response with your questions and proposals.

I suggest we would like responses by 30th June 2008 in order to allow
the author to revise the document before the deadlines. Please note that
we are looking to make this decision on the list within this deadline
based on responses received, not leave it until the Dublin meeting.

Regards

Keith
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Thread-Topic: re: [Sip] REFER a target-refresh
Thread-Index: AcjR+BXg7Q/mXoppTLaHmP27W6FO9A==
From: "Rockson Li (zhengyli)" <zhengyli@cisco.com>
To: <christer.holmberg@ericsson.com>, <sip@ietf.org>
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Subject: Re: [Sip] REFER a target-refresh
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Hi Christer,

I think it's target refresh method. please check RFC5057


5.4. Target Refresh Requests

   Target refresh requests update the remote target of a dialog when
   they are successfully processed.  The currently defined target
   refresh requests are INVITE, UPDATE, SUBSCRIBE, NOTIFY, and REFER.

Regards,
-Rockson 

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

Hi, 

I have a question regarding the REFER method: is it a target refresh
method? 

RFC 3515 doesn't say anything about it. However, the Contact header
seems to be MANDATED in the REFER request (eventhough not shown in the
examples in the draft).

And, chapter 4 of RFC 4488 says: 

"This document defines a new SIP header field: "Refer-Sub". This header
field is meaningful and MAY be used with a REFER request and the
corresponding 2XX response only. This header field set to "false"
specifies that a REFER-Issuer requests that the REFER- Recipient doesn't
establish an implicit subscription and the resultant dialog. Note that
when using this extension, the REFER remains a target refresh request
(as in the default case -- when the extension is not used)."

So, I guess that means that REFER IS a target refresh method? 

Regards, 

Christer 

_______________________________________________
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Use sip-implementors@cs.columbia.edu for questions on current sip
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Thread-Topic: re: [Sip] REFER a target-refresh
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From: "Rockson Li (zhengyli)" <zhengyli@cisco.com>
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Subject: Re: [Sip] REFER a target-refresh
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And more http://bugs.sipit.net/show_bug.cgi?id=764 

-----Original Message-----
From: Rockson Li (zhengyli) 
Sent: Thursday, June 19, 2008 6:35 PM
To: 'christer.holmberg@ericsson.com'; 'sip@ietf.org'
Subject: re: [Sip] REFER a target-refresh

Hi Christer,

I think it's target refresh method. please check RFC5057


5.4. Target Refresh Requests

   Target refresh requests update the remote target of a dialog when
   they are successfully processed.  The currently defined target
   refresh requests are INVITE, UPDATE, SUBSCRIBE, NOTIFY, and REFER.

Regards,
-Rockson 

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

Hi, 

I have a question regarding the REFER method: is it a target refresh
method? 

RFC 3515 doesn't say anything about it. However, the Contact header
seems to be MANDATED in the REFER request (eventhough not shown in the
examples in the draft).

And, chapter 4 of RFC 4488 says: 

"This document defines a new SIP header field: "Refer-Sub". This header
field is meaningful and MAY be used with a REFER request and the
corresponding 2XX response only. This header field set to "false"
specifies that a REFER-Issuer requests that the REFER- Recipient doesn't
establish an implicit subscription and the resultant dialog. Note that
when using this extension, the REFER remains a target refresh request
(as in the default case -- when the extension is not used)."

So, I guess that means that REFER IS a target refresh method? 

Regards, 

Christer 

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


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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
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Thanks!

Regards,

Christer =


-----Original Message-----
From: Rockson Li (zhengyli) [mailto:zhengyli@cisco.com] =

Sent: 19. kes=E4kuuta 2008 13:35
To: Christer Holmberg; sip@ietf.org
Subject: re: [Sip] REFER a target-refresh

Hi Christer,

I think it's target refresh method. please check RFC5057


5.4. Target Refresh Requests

   Target refresh requests update the remote target of a dialog when
   they are successfully processed.  The currently defined target
   refresh requests are INVITE, UPDATE, SUBSCRIBE, NOTIFY, and REFER.

Regards,
-Rockson =


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

Hi, =


I have a question regarding the REFER method: is it a target refresh method=
? =


RFC 3515 doesn't say anything about it. However, the Contact header seems t=
o be MANDATED in the REFER request (eventhough not shown in the examples in=
 the draft).

And, chapter 4 of RFC 4488 says: =


"This document defines a new SIP header field: "Refer-Sub". This header fie=
ld is meaningful and MAY be used with a REFER request and the corresponding=
 2XX response only. This header field set to "false"
specifies that a REFER-Issuer requests that the REFER- Recipient doesn't es=
tablish an implicit subscription and the resultant dialog. Note that when u=
sing this extension, the REFER remains a target refresh request (as in the =
default case -- when the extension is not used)."

So, I guess that means that REFER IS a target refresh method? =


Regards, =


Christer =


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3261 says

Specifically, requests that are not target refresh requests do not
modify the dialog's remote target URI, and requests that are target
refresh requests do. For dialogs that have been established with an
INVITE, the only target refresh request defined is re-INVITE (see
Section 14). Other extensions may define different target refresh
requests for dialogs established in other ways. 

REFER does change the remote-target URI.
 
Best Regards,
Ashish Saxena
877-5570

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Rockson Li (zhengyli)
Sent: Thursday, June 19, 2008 4:05 PM
To: christer.holmberg@ericsson.com; sip@ietf.org
Subject: Re: [Sip] REFER a target-refresh

Hi Christer,

I think it's target refresh method. please check RFC5057


5.4. Target Refresh Requests

   Target refresh requests update the remote target of a dialog when
   they are successfully processed.  The currently defined target
   refresh requests are INVITE, UPDATE, SUBSCRIBE, NOTIFY, and REFER.

Regards,
-Rockson 

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

Hi, 

I have a question regarding the REFER method: is it a target refresh
method? 

RFC 3515 doesn't say anything about it. However, the Contact header
seems to be MANDATED in the REFER request (eventhough not shown in the
examples in the draft).

And, chapter 4 of RFC 4488 says: 

"This document defines a new SIP header field: "Refer-Sub". This header
field is meaningful and MAY be used with a REFER request and the
corresponding 2XX response only. This header field set to "false"
specifies that a REFER-Issuer requests that the REFER- Recipient doesn't
establish an implicit subscription and the resultant dialog. Note that
when using this extension, the REFER remains a target refresh request
(as in the default case -- when the extension is not used)."

So, I guess that means that REFER IS a target refresh method? 

Regards, 

Christer 

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


From sip-bounces@ietf.org  Thu Jun 19 10:38:26 2008
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From: "Milan Patel" <milanpa@nortel.com>
To: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>, <sip@ietf.org>
Subject: Re: [Sip] draft-dotson-sip-mutual-auth-03
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Hi,

I too would like to show support for the draft being progressed as a WG ite=
m. The procedures in the draft are indeed useful to 3GPP and the Proxy-Auth=
entication-Info header has been included in 3GPP specifications. The additi=
on of Motivation in section 3 is a useful addition to the draft.

Regards,
Milan

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Chris=
ter Holmberg
Sent: 16 June 2008 12:43
To: DRAGE, Keith (Keith); sip@ietf.org
Subject: Re: [Sip] draft-dotson-sip-mutual-auth-03


Hi,

My position of the draft is supportive.

Regards,

Christer
 =


-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of DRAGE=
, Keith (Keith)
Sent: 12. kes=E4kuuta 2008 16:50
To: sip@ietf.org
Subject: Re: [Sip] draft-dotson-sip-mutual-auth-03

(As SIP WG cochair)

The document has now been updated based on comments received.

http://www.ietf.org/internet-drafts/draft-dotson-sip-mutual-auth-03.txt

I will give a few more days for responses to the question of support / no s=
upport.

Please read the draft and indicate your postion.

Regards

Keith =


> -----Original Message-----
> From: Stuart Hoggan [mailto:s.hoggan@CableLabs.com]
> Sent: Tuesday, June 10, 2008 11:01 PM
> To: DRAGE, Keith (Keith); sip@ietf.org
> Subject: RE: [Sip] draft-dotson-sip-mutual-auth-02
> =

> All,
> =

> We have updated the document to incorporate feedback received so far.
> It now contains a section that helps clarify the motivation.  =

> Let us know if you have further comments. =

> =

> -Stuart
> =

> > -----Original Message-----
> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
> Behalf Of
> > DRAGE, Keith (Keith)
> > Sent: Wednesday, June 04, 2008 3:48 AM
> > To: sip@ietf.org
> > Subject: Re: [Sip] draft-dotson-sip-mutual-auth-02
> > =

> > (As SIP WG cochair)
> > =

> > This has been raised in the SIP group a couple of times, and we have
> not
> > yet gained an idea of whether to proceed with it in the SIP
> WG or not.
> > =

> > There have been a couple of technical comments raised in
> the past from
> > the security experts; my understanding is that these have now been =

> > clarified.
> > =

> > I believe in the past there has been an interested
> community that says
> > this is useful. It has also been clarified that 3GPP would
> like to use
> > it in their specifications.
> > =

> > So could I ask the WG to look at this document, and formally to
> indicate
> > whether they believe some work with this scope should be
> progressed as
> a
> > SIP WG item.
> > =

> >
> http://www.ietf.org/internet-drafts/draft-dotson-sip-mutual-au
> th-02.txt
> > =

> > So if you support it please indicate to the SIP WG chairs.
> > =

> > If you have technical concerns (i.e. it should not proceed
> because it
> is
> > technically flawed beyond fixing) then please ideally mail
> the list,
> > although the SIP WG chairs will also accept input.
> > =

> > Responses please in 1 calendar week from this date.
> > =

> > Regards
> > =

> > Keith
> > =

> > =

> > ________________________________
> > =

> > 	From: sip-bounces@ietf.org
> [mailto:sip-bounces@ietf.org] On Behalf Of
> > Stuart Hoggan
> > 	Sent: Friday, May 02, 2008 5:15 PM
> > 	To: sip@ietf.org
> > 	Subject: [Sip] draft-dotson-sip-mutual-auth-02
> > =

> > =

> > =

> > 	Folks,
> > =

> > =

> > =

> > 	Based on the feedback received in Philadelphia, we have updated =

> > draft-dotson-sip-mutual-auth. The changes clarify the usage of the =

> > Proxy-Authentication-Info header, including support for multiple
> values.
> > =

> > =

> > =

> > =

> > 	Any further comments or suggestions will be
> appreciated. It would
> > also be nice to obtain WG feedback on pursuing this I-D as a WG =

> > document.
> > =

> > =

> > =

> > 	Regards,
> > =

> > 	Stuart
> > =

> > =

> > =

> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use =

> > sip-implementors@cs.columbia.edu for questions on current sip Use =

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

_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol Use sip-implement=
ors@cs.columbia.edu for questions on current sip Use sipping@ietf.org for n=
ew developments on the application of sip
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
_______________________________________________
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This list is for NEW development of the core SIP 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: "DOLLY, MARTIN C, ATTLABS" <mdolly@att.com>
To: "Milan Patel" <milanpa@nortel.com>,
	"DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>, <sip@ietf.org>
Subject: Re: [Sip] draft-dotson-sip-mutual-auth-03
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Keith,

We are supportive of this draft.

Cheers,

Martin =


-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Milan=
 Patel
Sent: Thursday, June 19, 2008 1:39 PM
To: DRAGE, Keith (Keith); sip@ietf.org
Subject: Re: [Sip] draft-dotson-sip-mutual-auth-03

Hi,

I too would like to show support for the draft being progressed as a WG ite=
m. The procedures in the draft are indeed useful to 3GPP and the Proxy-Auth=
entication-Info header has been included in 3GPP specifications. The additi=
on of Motivation in section 3 is a useful addition to the draft.

Regards,
Milan

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Chris=
ter Holmberg
Sent: 16 June 2008 12:43
To: DRAGE, Keith (Keith); sip@ietf.org
Subject: Re: [Sip] draft-dotson-sip-mutual-auth-03


Hi,

My position of the draft is supportive.

Regards,

Christer
 =


-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of DRAGE=
, Keith (Keith)
Sent: 12. kes=E4kuuta 2008 16:50
To: sip@ietf.org
Subject: Re: [Sip] draft-dotson-sip-mutual-auth-03

(As SIP WG cochair)

The document has now been updated based on comments received.

http://www.ietf.org/internet-drafts/draft-dotson-sip-mutual-auth-03.txt

I will give a few more days for responses to the question of support / no s=
upport.

Please read the draft and indicate your postion.

Regards

Keith =


> -----Original Message-----
> From: Stuart Hoggan [mailto:s.hoggan@CableLabs.com]
> Sent: Tuesday, June 10, 2008 11:01 PM
> To: DRAGE, Keith (Keith); sip@ietf.org
> Subject: RE: [Sip] draft-dotson-sip-mutual-auth-02
> =

> All,
> =

> We have updated the document to incorporate feedback received so far.
> It now contains a section that helps clarify the motivation.  =

> Let us know if you have further comments. =

> =

> -Stuart
> =

> > -----Original Message-----
> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
> Behalf Of
> > DRAGE, Keith (Keith)
> > Sent: Wednesday, June 04, 2008 3:48 AM
> > To: sip@ietf.org
> > Subject: Re: [Sip] draft-dotson-sip-mutual-auth-02
> > =

> > (As SIP WG cochair)
> > =

> > This has been raised in the SIP group a couple of times, and we have
> not
> > yet gained an idea of whether to proceed with it in the SIP
> WG or not.
> > =

> > There have been a couple of technical comments raised in
> the past from
> > the security experts; my understanding is that these have now been =

> > clarified.
> > =

> > I believe in the past there has been an interested
> community that says
> > this is useful. It has also been clarified that 3GPP would
> like to use
> > it in their specifications.
> > =

> > So could I ask the WG to look at this document, and formally to
> indicate
> > whether they believe some work with this scope should be
> progressed as
> a
> > SIP WG item.
> > =

> >
> http://www.ietf.org/internet-drafts/draft-dotson-sip-mutual-au
> th-02.txt
> > =

> > So if you support it please indicate to the SIP WG chairs.
> > =

> > If you have technical concerns (i.e. it should not proceed
> because it
> is
> > technically flawed beyond fixing) then please ideally mail
> the list,
> > although the SIP WG chairs will also accept input.
> > =

> > Responses please in 1 calendar week from this date.
> > =

> > Regards
> > =

> > Keith
> > =

> > =

> > ________________________________
> > =

> > 	From: sip-bounces@ietf.org
> [mailto:sip-bounces@ietf.org] On Behalf Of
> > Stuart Hoggan
> > 	Sent: Friday, May 02, 2008 5:15 PM
> > 	To: sip@ietf.org
> > 	Subject: [Sip] draft-dotson-sip-mutual-auth-02
> > =

> > =

> > =

> > 	Folks,
> > =

> > =

> > =

> > 	Based on the feedback received in Philadelphia, we have updated =

> > draft-dotson-sip-mutual-auth. The changes clarify the usage of the =

> > Proxy-Authentication-Info header, including support for multiple
> values.
> > =

> > =

> > =

> > =

> > 	Any further comments or suggestions will be
> appreciated. It would
> > also be nice to obtain WG feedback on pursuing this I-D as a WG =

> > document.
> > =

> > =

> > =

> > 	Regards,
> > =

> > 	Stuart
> > =

> > =

> > =

> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use =

> > sip-implementors@cs.columbia.edu for questions on current sip Use =

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

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


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Hi Brett -

I got to go through this very carefully today. You found a real bug  
here.
Timer D is going to have to be set to 64*T1 without regard to transport.
It needs to balance what's potentially happening on the server side with
either Timer H or Timer L.

I'll also go through the whole text again to see if this was an  
isolated bug
or a class-of bug that has other instances, and will issue an update  
to the
draft shortly.

RjS

On May 27, 2008, at 10:01 AM, Brett Tate wrote:

> Thanks for the response.  Reply inline.
>
>>> Section 8.4:
>>>
>>> Concerning timer D and "zero seconds for reliable
>>> transports", is "reliable" obtained from INVITE
>>> or ACK.
>>>
>>> Concerning timer D being zero, at first glance
>>> it looks like the state machine text no longer
>>> ensures ACK will be sent for extra 2xx or
>>> 300-699.  The same applies, to ensuring ACK resent
>>> in situations where transport not reliable
>>> end-to-end.  Per 7.2 paragraph 2, this might be
>>> intentional; if so, the motivation and flexibility
>>> in handling from section 7.2 should potentially
>>> also be included within 8.4.  If I'm misinterpreting
>>> the fix, where does it say the ACK MUST, SHOULD, or
>>> MAY still be (re-)sent for the extra responses when
>>> timer D is 0?
>>
>> Well, the main requirement you're looking for is
>> (still) on the TU.  But I see your point in wondering
>> if a retransmitted 200 will still  _get_ to the TU if
>> it came over UDP. Let me look at it for awhile.
>
> Okay.
>
>
>> To carve that away from some of the other questions
>> you ask above:
>>
>> You're talking about receiving a response over a
>> reliable transport  (like TCP).
>>
>> Over such a transport, there won't _be_ any
>> additional 300-699 responses unless the thing on the
>> other end is violating the specification - remember
>> those are hop-by-hop messages, and the thing sending
>> them is required to hand reliability to the reliable
>> transport (it won't retransmit - timer G is not set
>> for reliable transports).  The path through the machines
>> when a 300-699 occurs hasn't changed from 3261.
>
> By extra 300-699, I was mainly referring to 1) race conditions with  
> 2xx
> and 2) interoperability with pre-rfc3261 devices.
>
> 1) This should be rare since usually only results from connection
> issues, threading issues, or stateless proxy issues causing the
> reordering.
>
> 2) Unfortunately this is not as rare as everyone would like.   
> However it
> is likely not much of an issue if the invfix chooses not to remain
> backwards compatible with rfc2543 concerning resending the ACK.
>
>
>> For a 200 class response, the transaction state machine doesn't send
>> the ACK - that is still the TU's job and this draft doesn't change
>> that. What we need to make sure is that we're leaving the path for a
>> retransmitted (or a 200 from another branch) to _get_ to the TU. I
>> thought we had, but its possible that's messed up. I'll look at it
>> some more.
>
> Okay.

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Hrm.

While working this over, it occurs to me that this might not be the  
right thing to do.

It would force the client transaction to continue to exist for 32  
seconds(ish) after acknowledging
a >=300 response over TCP. There's no reason to do so in this case  
since there will never be
a retransmission of the response to absorb (failure responses are hop- 
by-hop and the hop is TCP,
so this client would not have retransmitted the request).

It probably makes more sense to use another state in the client,  
matching the new state in the server
instead of just trying to reuse this existing state. This new state  
would be where the client transaction
sits waiting for multiple (or retransmitted) 2xx responses. The  
Completed state would, then, be
completely unchanged from what's in 3261.

Anyhow, the next rev will have a concrete proposal.

RjS

On Jun 19, 2008, at 4:42 PM, Robert Sparks wrote:

> Hi Brett -
>
> I got to go through this very carefully today. You found a real bug  
> here.
> Timer D is going to have to be set to 64*T1 without regard to  
> transport.
> It needs to balance what's potentially happening on the server side  
> with
> either Timer H or Timer L.
>
> I'll also go through the whole text again to see if this was an  
> isolated bug
> or a class-of bug that has other instances, and will issue an update  
> to the
> draft shortly.
>
> RjS
>
> On May 27, 2008, at 10:01 AM, Brett Tate wrote:
>
>> Thanks for the response.  Reply inline.
>>
>>>> Section 8.4:
>>>>
>>>> Concerning timer D and "zero seconds for reliable
>>>> transports", is "reliable" obtained from INVITE
>>>> or ACK.
>>>>
>>>> Concerning timer D being zero, at first glance
>>>> it looks like the state machine text no longer
>>>> ensures ACK will be sent for extra 2xx or
>>>> 300-699.  The same applies, to ensuring ACK resent
>>>> in situations where transport not reliable
>>>> end-to-end.  Per 7.2 paragraph 2, this might be
>>>> intentional; if so, the motivation and flexibility
>>>> in handling from section 7.2 should potentially
>>>> also be included within 8.4.  If I'm misinterpreting
>>>> the fix, where does it say the ACK MUST, SHOULD, or
>>>> MAY still be (re-)sent for the extra responses when
>>>> timer D is 0?
>>>
>>> Well, the main requirement you're looking for is
>>> (still) on the TU.  But I see your point in wondering
>>> if a retransmitted 200 will still  _get_ to the TU if
>>> it came over UDP. Let me look at it for awhile.
>>
>> Okay.
>>
>>
>>> To carve that away from some of the other questions
>>> you ask above:
>>>
>>> You're talking about receiving a response over a
>>> reliable transport  (like TCP).
>>>
>>> Over such a transport, there won't _be_ any
>>> additional 300-699 responses unless the thing on the
>>> other end is violating the specification - remember
>>> those are hop-by-hop messages, and the thing sending
>>> them is required to hand reliability to the reliable
>>> transport (it won't retransmit - timer G is not set
>>> for reliable transports).  The path through the machines
>>> when a 300-699 occurs hasn't changed from 3261.
>>
>> By extra 300-699, I was mainly referring to 1) race conditions with  
>> 2xx
>> and 2) interoperability with pre-rfc3261 devices.
>>
>> 1) This should be rare since usually only results from connection
>> issues, threading issues, or stateless proxy issues causing the
>> reordering.
>>
>> 2) Unfortunately this is not as rare as everyone would like.   
>> However it
>> is likely not much of an issue if the invfix chooses not to remain
>> backwards compatible with rfc2543 concerning resending the ACK.
>>
>>
>>> For a 200 class response, the transaction state machine doesn't send
>>> the ACK - that is still the TU's job and this draft doesn't change
>>> that. What we need to make sure is that we're leaving the path for a
>>> retransmitted (or a 200 from another branch) to _get_ to the TU. I
>>> thought we had, but its possible that's messed up. I'll look at it
>>> some more.
>>
>> Okay.
>
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Hi Christer,
	From Section 4 (Client Behaviour) part of this draft, I
understood that primary objective of this draft is to give indication to
UAC to release resources, if any. I am little worried about extra
messaging that will be seen because of 199 implementation. Can there be
a way to tell proxies and UAS downstream that UAC has some resources
reserved for this call and would be interested in receiving 199, if
possible?

IMO, UAC (mostly) is in better understanding of the resources that it
has reserved for a particular call. So UAC can selectively add something
in INVITE to tell downstream proxies/ UAS' about its interest. 

IMO, we can have a package for 199 and UAC can use "Supported" header to
indicate entities downstream that it would be interested in getting 199
for this *particular* call. Implicitly it would mean that UAC has
reserved some resources. I understand that we would changing the meaning
of "Supported" header on per call basis :-(. Something else may be?

Comments please.

Best Regards,
Ashish Saxena
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Hi,

  I have a question regarding provisional responses in SIP. Can the username/displayname part of To/From header be changed in a provisional response to a SIP request ? If so, can you please direct me to a draft/rfc which states this can be done ?


Thanks and Regards,
Brijesh 


      
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At 01:10 AM 6/20/2008, brijesh nair wrote:
>Hi,
>
>   I have a question regarding provisional responses in SIP. Can the 
> username/displayname part of To/From header be changed in a 
> provisional response to a SIP request ? If so, can you please 
> direct me to a draft/rfc which states this can be done ?

RFC 4474 (SIP Identity) states the display name in the To header 
cannot be prevented from being changed (therefore shouldn't be 
trusted) in any response.  Does that answer your question?



>Thanks and Regards,
>Brijesh
>
>
>
>_______________________________________________
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>Use sip-implementors@cs.columbia.edu for questions on current sip
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Thread-Topic: Re: [Sip] Regarding provisional responses to SIP requests
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From: "Rockson Li (zhengyli)" <zhengyli@cisco.com>
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Hi  Brijesh,
 
For RFC3261-complianet UA, this can be done, the main concern over this
modification is dialog matching.
RFC2543 use whole from/to/call-id header to match dialog, whereas,
RFC3261 uses only from tag / to tag/call-id
 
So we can find in RFC sec5.3

   Currently, dialog matching uses only the
   tags in the To and From headers, rather than the whole header fields.
   Thus, under the new rules the URI values in the To and From headers
   themselves could be altered by intermediaries.  However, some legacy
   clients might consider it an error condition if the value of the URI
   in the From header altered between the request and the response.
 
But to better cover RFC2543 UA it later says,

   Therefore, any time that a privacy service needs to modify any
   dialog-matching headers for privacy reasons, it SHOULD act as a
   transparent back-to-back user agent, and it MUST persist the former
   values of the dialog-matching headers.  These values MUST be restored
   in any messages that are sent to the originating user agent.

So, IMO, if the UAs you are facing is based on RFC3261, I think the from
header change is OK.

Regards,
-Rockson 

On Fri, Jun 20, 2008 at  2:55 PM, James M. Polk wrote:
 
> At 01:10 AM 6/20/2008, brijesh nair wrote:
> > Hi,
> > 
> >   I have a question regarding provisional responses in SIP. Can the 
> > username/displayname part of To/From header be changed in a 
> > provisional response to a SIP request ? If so, can you please 
> > direct me to a draft/rfc which states this can be done ?
> 
> RFC 4474 (SIP Identity) states the display name in the To header
cannot be prevented from being changed (therefore shouldn't be trusted)
in any response.  Does that answer your question?
> 
> 
> 
> > Thanks and Regards,
> > Brijesh
> > 
> > 
> > 
> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


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Thread-Topic: Re: [Sip] Regarding provisional responses to SIP requests
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From: "Rockson Li (zhengyli)" <zhengyli@cisco.com>
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Subject: Re: [Sip] Regarding provisional responses to SIP requests
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Forget the RFC id , RFC3323
-----Original Message-----
From: Rockson Li (zhengyli) 
Sent: Friday, June 20, 2008 5:04 PM
To: 'briju31@yahoo.com'
Cc: 'sip@ietf.org'
Subject: Re: [Sip] Regarding provisional responses to SIP requests

Hi  Brijesh,
 
For RFC3261-complianet UA, this can be done, the main concern over this
modification is dialog matching.
RFC2543 use whole from/to/call-id header to match dialog, whereas,
RFC3261 uses only from tag / to tag/call-id
 
So we can find in RFC sec5.3
[Rockson]RFC3323

   Currently, dialog matching uses only the
   tags in the To and From headers, rather than the whole header fields.
   Thus, under the new rules the URI values in the To and From headers
   themselves could be altered by intermediaries.  However, some legacy
   clients might consider it an error condition if the value of the URI
   in the From header altered between the request and the response.
 
But to better cover RFC2543 UA it later says,

   Therefore, any time that a privacy service needs to modify any
   dialog-matching headers for privacy reasons, it SHOULD act as a
   transparent back-to-back user agent, and it MUST persist the former
   values of the dialog-matching headers.  These values MUST be restored
   in any messages that are sent to the originating user agent.

So, IMO, if the UAs you are facing is based on RFC3261, I think the from
header change is OK.

Regards,
-Rockson 

On Fri, Jun 20, 2008 at  2:55 PM, James M. Polk wrote:
 
> At 01:10 AM 6/20/2008, brijesh nair wrote:
> > Hi,
> > 
> >   I have a question regarding provisional responses in SIP. Can the 
> > username/displayname part of To/From header be changed in a 
> > provisional response to a SIP request ? If so, can you please direct

> > me to a draft/rfc which states this can be done ?
> 
> RFC 4474 (SIP Identity) states the display name in the To header
cannot be prevented from being changed (therefore shouldn't be trusted)
in any response.  Does that answer your question?
> 
> 
> 
> > Thanks and Regards,
> > Brijesh
> > 
> > 
> > 
> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use 
> > sip-implementors@cs.columbia.edu for questions on current sip Use 
> > sipping@ietf.org for new developments on the application of sip
> 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use 
> sip-implementors@cs.columbia.edu for questions on current sip Use 
> sipping@ietf.org for new developments on the application of sip
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>, "sip@ietf.org"
	<sip@ietf.org>
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My 2 cents: it is a useful draft.  Personally, I would like to have it available for two reasons not cited: PBX connections a la SIP trunks, and proxy-proxy connections.  Today I think a lot of people are using OPTIONS requests or proprietary means to perform such keep-alives when registration is not appropriate, but it has led to some interop issues and performance concerns in some cases.

Since the contentious issue of what form the keep-alives take have already been agreed on for outbound, this seems like a simple draft to get done. (famous last words, I know)

-hadriel


> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> DRAGE, Keith (Keith)
> Sent: Thursday, June 19, 2008 5:31 AM
> To: sip@ietf.org
> Cc: Christer Holmberg
> Subject: [Sip] Progress draft-holmberg-sip-keep
>
> (As SIP WG cochair)
>
> We have been asked by the author of
>
> http://www.ietf.org/internet-drafts/draft-holmberg-sip-keep-01.txt
>
> Whether the SIP WG can progress this document.
>
> Because this draft arose as a result of the discussion of outbound, and
> indeed seems to reuse the requirements from outbound, and these
> requirements never really got handled in the SIPPING WG, it has been
> agreed with the SIPPING chairs that we will handle this entirely within
> SIP.
>
> Now in order to ask for charter milestones, and indeed when we finally
> present this to IESG, we will be asked for the level of support in the
> WG, which is also predicated on does this fix a real problem, or is it
> just a corner case with limited application. So:
>
> QUESTION 1 TO SIP WG: Are the use cases sufficiently important to
> proceed with this draft? The document states:
>
>    Chapter 3.5 of draft-ietf-sip-outbound-13 [I-D.ietf-sip-outbound]
>    defines two keep-alive techniques.  Even though the keep-alive
>    techniques are separated from the Outbound mechanism
>    [I-D.ietf-sip-outbound], it is currently not possible to indicate
>    support of the keep-alive techniques without also indicating support
>    for the Outbound mechanism.
>
>    The Outbound mechanism is enabled during the UA registration phase.
>    However, there are use-cases where the UA does not register itself,
>    but still needs to be able to make calls and maintain NAT bindings
>    open during the duration of that call.  A typical example is
>    emergency calls.  There are also cases where entities do not support
>    the Outbound mechanism, but still want to be able to indicate support
>    and use the keep-alive techniques defined in [I-D.ietf-sip-outbound].
>
> At first sight this is not the most inspiring declaration of the need
> for the document. Please respond indicating whether you consider this a
> useful draft, and propose text that you think would be useful in this
> section. Conversely, if you think this draft is not useful and the WG
> has other more important things to work on first, please also respond.
>
> QUESTION 2 TO SIP WG: Do we have a robust set of requirements for
> proceeding with this work? The document currently lists:
>
>    REQ 1: It MUST be possible for a UA to indicate support of the keep-
>    alive techniques defined [I-D.ietf-sip-outbound] if the UA supports
>    only the keep-alive part of [I-D.ietf-sip-outbound].
>
>    REQ 2: It MUST be possible for an edge proxy to indicate support of
>    the keep-alive techniques defined [I-D.ietf-sip-outbound] if the edge
>    poxy supports only the keep-alive part of [I-D.ietf-sip-outbound].
>
> It would be desirable to agree these at the outset, and not revisit them
> if we continue with the work. So if you require clarification,
> modification, or addition to these two requirements, then please also
> response with your questions and proposals.
>
> I suggest we would like responses by 30th June 2008 in order to allow
> the author to revise the document before the deadlines. Please note that
> we are looking to make this decision on the list within this deadline
> based on responses received, not leave it until the Dublin meeting.
>
> Regards
>
> Keith
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


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Same here.

-hadriel

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> DOLLY, MARTIN C, ATTLABS
> Sent: Thursday, June 19, 2008 3:12 PM
> To: Milan Patel; DRAGE, Keith (Keith); sip@ietf.org
> Subject: Re: [Sip] draft-dotson-sip-mutual-auth-03
>
> Keith,
>
> We are supportive of this draft.
>
> Cheers,
>
> Martin
>
> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Milan Patel
> Sent: Thursday, June 19, 2008 1:39 PM
> To: DRAGE, Keith (Keith); sip@ietf.org
> Subject: Re: [Sip] draft-dotson-sip-mutual-auth-03
>
> Hi,
>
> I too would like to show support for the draft being progressed as a WG
> item. The procedures in the draft are indeed useful to 3GPP and the Proxy-
> Authentication-Info header has been included in 3GPP specifications. The
> addition of Motivation in section 3 is a useful addition to the draft.
>
> Regards,
> Milan
>
> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Christer Holmberg
> Sent: 16 June 2008 12:43
> To: DRAGE, Keith (Keith); sip@ietf.org
> Subject: Re: [Sip] draft-dotson-sip-mutual-auth-03
>
>
> Hi,
>
> My position of the draft is supportive.
>
> Regards,
>
> Christer
>
>
> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> DRAGE, Keith (Keith)
> Sent: 12. kes=E4kuuta 2008 16:50
> To: sip@ietf.org
> Subject: Re: [Sip] draft-dotson-sip-mutual-auth-03
>
> (As SIP WG cochair)
>
> The document has now been updated based on comments received.
>
> http://www.ietf.org/internet-drafts/draft-dotson-sip-mutual-auth-03.txt
>
> I will give a few more days for responses to the question of support / no
> support.
>
> Please read the draft and indicate your postion.
>
> Regards
>
> Keith
>
> > -----Original Message-----
> > From: Stuart Hoggan [mailto:s.hoggan@CableLabs.com]
> > Sent: Tuesday, June 10, 2008 11:01 PM
> > To: DRAGE, Keith (Keith); sip@ietf.org
> > Subject: RE: [Sip] draft-dotson-sip-mutual-auth-02
> >
> > All,
> >
> > We have updated the document to incorporate feedback received so far.
> > It now contains a section that helps clarify the motivation.
> > Let us know if you have further comments.
> >
> > -Stuart
> >
> > > -----Original Message-----
> > > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
> > Behalf Of
> > > DRAGE, Keith (Keith)
> > > Sent: Wednesday, June 04, 2008 3:48 AM
> > > To: sip@ietf.org
> > > Subject: Re: [Sip] draft-dotson-sip-mutual-auth-02
> > >
> > > (As SIP WG cochair)
> > >
> > > This has been raised in the SIP group a couple of times, and we have
> > not
> > > yet gained an idea of whether to proceed with it in the SIP
> > WG or not.
> > >
> > > There have been a couple of technical comments raised in
> > the past from
> > > the security experts; my understanding is that these have now been
> > > clarified.
> > >
> > > I believe in the past there has been an interested
> > community that says
> > > this is useful. It has also been clarified that 3GPP would
> > like to use
> > > it in their specifications.
> > >
> > > So could I ask the WG to look at this document, and formally to
> > indicate
> > > whether they believe some work with this scope should be
> > progressed as
> > a
> > > SIP WG item.
> > >
> > >
> > http://www.ietf.org/internet-drafts/draft-dotson-sip-mutual-au
> > th-02.txt
> > >
> > > So if you support it please indicate to the SIP WG chairs.
> > >
> > > If you have technical concerns (i.e. it should not proceed
> > because it
> > is
> > > technically flawed beyond fixing) then please ideally mail
> > the list,
> > > although the SIP WG chairs will also accept input.
> > >
> > > Responses please in 1 calendar week from this date.
> > >
> > > Regards
> > >
> > > Keith
> > >
> > >
> > > ________________________________
> > >
> > >     From: sip-bounces@ietf.org
> > [mailto:sip-bounces@ietf.org] On Behalf Of
> > > Stuart Hoggan
> > >     Sent: Friday, May 02, 2008 5:15 PM
> > >     To: sip@ietf.org
> > >     Subject: [Sip] draft-dotson-sip-mutual-auth-02
> > >
> > >
> > >
> > >     Folks,
> > >
> > >
> > >
> > >     Based on the feedback received in Philadelphia, we have updated
> > > draft-dotson-sip-mutual-auth. The changes clarify the usage of the
> > > Proxy-Authentication-Info header, including support for multiple
> > values.
> > >
> > >
> > >
> > >
> > >     Any further comments or suggestions will be
> > appreciated. It would
> > > also be nice to obtain WG feedback on pursuing this I-D as a WG
> > > document.
> > >
> > >
> > >
> > >     Regards,
> > >
> > >     Stuart
> > >
> > >
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol Use
> > > sip-implementors@cs.columbia.edu for questions on current sip Use
> > > sipping@ietf.org for new developments on the application of sip
> >
> _______________________________________________
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> This list is for NEW development of the core SIP Protocol Use sip-
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Henry Sinnreich <hsinnrei@adobe.com>, Dean Willis
	<dean.willis@softarmor.com>, "sip@ietf.org" <sip@ietf.org>
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> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Henry Sinnreich
>
> 3. The main deliverable of a WG is free software to validate and support
> the
> protocol document and its refinement. If no free software is contributed,
> the WG activity should be suspended. Internet-Drafts not supported by
> freely
> available software (that can be inspected and tried out) and measurements
> cannot be proven right or wrong and should therefore be ignored.
> There is plenty of IEEE and University work to prove this approach, which
> was also the basis for the rise and success of the IETF.

Since when is the main deliverable of the IETF free software?

The main deliverable of the IETF is documentation, for the purpose of achieving interoperability, to make the Internet better.  That's it. (i.e., RFC 3935)  Open source implementation is one means to an end goal, not the end goal itself.

And as it is, I think some of the extensions in SIP have *only* been implemented in open source software and never seen the light of day in real world use even of that open source software. (or in any degree one would consider "broad use")  In that sense, maybe we should only continue work in things that have seen commercial implementation! ;)

-hadriel
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From sip-bounces@ietf.org  Fri Jun 20 11:16:38 2008
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Dean Willis <dean.willis@softarmor.com>, "sip@ietf.org" <sip@ietf.org>
Date: Fri, 20 Jun 2008 14:15:48 -0400
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Subject: Re: [Sip] INFO and what to do about it?
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[Not surprisingly] I also think we should just do it.  I don't know if it will succeed, but the longer we wait the less the probability it will.  Right now people have to provision devices to send XYZ thing in INFO, and different vendors are using different syntaxes/means for the same mechanism, and I loathe that.  If we want to stop devices in the middle from meddling, we need to stop giving them reasons to.

-hadriel

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Dean
> Willis
> Sent: Thursday, June 19, 2008 12:09 AM
> To: sip@ietf.org
> Cc: Cullen Jennings; Jon Peterson
> Subject: Re: [Sip] INFO and what to do about it?
>
> I asked:
>
>
> > We've been batting INFO around for many, many years now (like, 10 --
> > it predates the SIP working group). A couple of meetings back, we
> > agreed that we would discuss use cases for INFO packages, and if we
> > didn't find any consensus there, then we would go ahead and publish an
> > "INFO Considered Harmful" RFC.
> >
> > We tried having this discussion at the last IETF, but that just didn't
> > work out.
> >
> > Do we try again, or just give up and publish "INFO Considered
> > Harmful"?
> >
> > Personally, I don't care anymore -- I just want to drive a stake into
> > the heart of this undead-thing, cut off its head, stuff the mouth with
> > holy wafers, and bury it at a crossroads somewhere.
> >
> > But this WG only slays by consensus, so what do you want?
>
> Several people (Jonathan, Mary, Christer, Paul) have responded to the
> list  with various things that added up to "Publish INFO packages AND
> document the extension models for SIP". So far, nobody has suggested
> otherwise.
>
> Does anybody want to do anything else? Speak now or forever visualize
> whirled peas.
>
> --
> Dean
>
>
> _______________________________________________
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I agree with Hadriel on this one.  I'm getting requests for different oddba=
ll uses of INFO all of the time, most of them driven by the need to interop=
erate with deployed applications.  Creating real INFO packages would give u=
s a way to increase the likelihood of interoperability for the different us=
es.  Is it too late?  Hard to say, but it is worth a shot.   =


James    =


James Rafferty =

Product Line Director, Integrated Media Gateways =

Dialogic, Inc.=A0 =

15 Crawford Street =

Needham, MA 02494 =

USA =


Tel:=A0=A0=A0 781 433 9462
Mobile: 781 929 3895
Fax:=A0=A0=A0 781 433 9268
Email:=A0 james.rafferty@dialogic.com
Web:=A0=A0=A0 www.dialogic.com
This e-mail is intended only for the named recipient(s) and may contain inf=
ormation that is privileged, confidential and/or exempt from disclosure und=
er applicable law. No waiver of privilege, confidence or otherwise is inten=
ded by virtue of communication via the internet. Any unauthorized use, diss=
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-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Hadri=
el Kaplan
Sent: Friday, June 20, 2008 2:16 PM
To: Dean Willis; sip@ietf.org
Cc: Cullen Jennings; Jon Peterson
Subject: Re: [Sip] INFO and what to do about it?


[Not surprisingly] I also think we should just do it.  I don't know if it w=
ill succeed, but the longer we wait the less the probability it will.  Righ=
t now people have to provision devices to send XYZ thing in INFO, and diffe=
rent vendors are using different syntaxes/means for the same mechanism, and=
 I loathe that.  If we want to stop devices in the middle from meddling, we=
 need to stop giving them reasons to.

-hadriel

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Dean
> Willis
> Sent: Thursday, June 19, 2008 12:09 AM
> To: sip@ietf.org
> Cc: Cullen Jennings; Jon Peterson
> Subject: Re: [Sip] INFO and what to do about it?
>
> I asked:
>
>
> > We've been batting INFO around for many, many years now (like, 10 --
> > it predates the SIP working group). A couple of meetings back, we
> > agreed that we would discuss use cases for INFO packages, and if we
> > didn't find any consensus there, then we would go ahead and publish an
> > "INFO Considered Harmful" RFC.
> >
> > We tried having this discussion at the last IETF, but that just didn't
> > work out.
> >
> > Do we try again, or just give up and publish "INFO Considered
> > Harmful"?
> >
> > Personally, I don't care anymore -- I just want to drive a stake into
> > the heart of this undead-thing, cut off its head, stuff the mouth with
> > holy wafers, and bury it at a crossroads somewhere.
> >
> > But this WG only slays by consensus, so what do you want?
>
> Several people (Jonathan, Mary, Christer, Paul) have responded to the
> list  with various things that added up to "Publish INFO packages AND
> document the extension models for SIP". So far, nobody has suggested
> otherwise.
>
> Does anybody want to do anything else? Speak now or forever visualize
> whirled peas.
>
> --
> Dean
>
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
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Hadriel Kaplan writes:

 > The main deliverable of the IETF is documentation, for the purpose of
 > achieving interoperability, to make the Internet better.  

please let me know how all those 3gpp/ims related drafts/rfcs that
sip[ping] wgs have produced help to make Internet better?

-- juha
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Thread-Topic: [Sip] Toward the Evolution of SIP and Related Working Groups
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If by "3GPP/IMS related drafts/rfcs" you mean the P-headers in 3455, it's the IETF that wanted them to be submitted I thought.  It's only informational.  What drafts were you thinking of?

They use the Internet, or their private networks connect to the Internet, and thus we should want them to tell us what they feel is missing from SIP.  We can use the info to decide if it's generally useful for everyone, or not, or even as an example of what not to do and try to persuade them down a different path.  They potentially represent a significant population of UA's that we should want to be reachable over the Internet in some form, to/from pure IETF-compliant UA's.

-hadriel

> -----Original Message-----
> From: Juha Heinanen [mailto:jh@tutpro.com]
> Sent: Friday, June 20, 2008 2:50 PM
> To: Hadriel Kaplan
> Cc: Henry Sinnreich; Dean Willis; sip@ietf.org
> Subject: Re: [Sip] Toward the Evolution of SIP and Related Working Groups
>
> Hadriel Kaplan writes:
>
>  > The main deliverable of the IETF is documentation, for the purpose of
>  > achieving interoperability, to make the Internet better.
>
> please let me know how all those 3gpp/ims related drafts/rfcs that
> sip[ping] wgs have produced help to make Internet better?
>
> -- juha
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


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> The main deliverable of the IETF is documentation, for the purpose of
> achieving interoperability, to make the Internet better.

Docs without Code
Documentation without available code has led us to use Jabber IM in the
SIMPLE WG meetings. Is this a success story for SIMPLE/SIP?

Serial Writers 
Other harmful effects of writing specs on the editor "platform" is the
deluge of SIP related RFCs and I-Ds since there is no threshold to stop
serial writers. 
* Does anyone expect developers to read and understand the 100's of
documents?
* How can the concepts and definitions in a document without code be proven
in the first place? By discussion and voting?

In Conflict With The Internet Principles
Last but not least, supporting commercial products or business plans is not
the same as supporting the Internet. There may be actually a fundamental
conflict here, for example closed stacks vs. the e2e principle, supporting
Internet transparency, etc.  It is the reason why all work related to the
service provider market, which is important work BTW, should be moved to the
ITU-T IMO.

These topics deserves however a larger discussion, outside of this list.

Enjoy the weekend,

Henry


On 6/20/08 1:02 PM, "Hadriel Kaplan" <HKaplan@acmepacket.com> wrote:

> 
> 
>> -----Original Message-----
>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
>> Henry Sinnreich
>> 
>> 3. The main deliverable of a WG is free software to validate and support
>> the
>> protocol document and its refinement. If no free software is contributed,
>> the WG activity should be suspended. Internet-Drafts not supported by
>> freely
>> available software (that can be inspected and tried out) and measurements
>> cannot be proven right or wrong and should therefore be ignored.
>> There is plenty of IEEE and University work to prove this approach, which
>> was also the basis for the rise and success of the IETF.
> 
> Since when is the main deliverable of the IETF free software?
> 
> The main deliverable of the IETF is documentation, for the purpose of
> achieving interoperability, to make the Internet better.  That's it. (i.e.,
> RFC 3935)  Open source implementation is one means to an end goal, not the end
> goal itself.
> 
> And as it is, I think some of the extensions in SIP have *only* been
> implemented in open source software and never seen the light of day in real
> world use even of that open source software. (or in any degree one would
> consider "broad use")  In that sense, maybe we should only continue work in
> things that have seen commercial implementation! ;)
> 
> -hadriel

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From: Cullen Jennings <fluffy@cisco.com>
To: Dean Willis <dean.willis@softarmor.com>
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Subject: Re: [Sip] INFO and what to do about it?
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Personally, I'd like to see some coherent discussion around it in a WG  
meeting then try to get to consensus on it. The totally number of  
people bothering to follow the conversation at this point is extremely  
low.

On Jun 18, 2008, at 10:08 PM, Dean Willis wrote:

> I asked:
>
>
> > We've been batting INFO around for many, many years now (like, 10 --
> > it predates the SIP working group). A couple of meetings back, we
> > agreed that we would discuss use cases for INFO packages, and if we
> > didn't find any consensus there, then we would go ahead and  
> publish an
> > "INFO Considered Harmful" RFC.
> >
> > We tried having this discussion at the last IETF, but that just  
> didn't
> > work out.
> >
> > Do we try again, or just give up and publish "INFO Considered
> > Harmful"?
> >
> > Personally, I don't care anymore -- I just want to drive a stake  
> into
> > the heart of this undead-thing, cut off its head, stuff the mouth  
> with
> > holy wafers, and bury it at a crossroads somewhere.
> >
> > But this WG only slays by consensus, so what do you want?
>
> Several people (Jonathan, Mary, Christer, Paul) have responded to the
> list  with various things that added up to "Publish INFO packages AND
> document the extension models for SIP". So far, nobody has suggested
> otherwise.
>
> Does anybody want to do anything else? Speak now or forever visualize
> whirled peas.
>
> --
> Dean
>
>

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


From sip-bounces@ietf.org  Sat Jun 21 12:10:41 2008
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To: SIP@ietf.org
From: eburger@standardstrack.com
Date: Sat, 21 Jun 2008 19:10:44 +0000
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Subject: Re: [Sip] INFO and what to do about it?
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If people care, I'll work, but if nobody cares, I won't.
Sent from my Verizon Wireless BlackBerry

-----Original Message-----
From: Cullen Jennings <fluffy@cisco.com>

Date: Thu, 19 Jun 2008 17:49:00 
To:Dean Willis <dean.willis@softarmor.com>
Cc:sip@ietf.org, Jon Peterson <jon.peterson@neustar.biz>
Subject: Re: [Sip] INFO and what to do about it?



Personally, I'd like to see some coherent discussion around it in a WG  
meeting then try to get to consensus on it. The totally number of  
people bothering to follow the conversation at this point is extremely  
low.

On Jun 18, 2008, at 10:08 PM, Dean Willis wrote:

> I asked:
>
>
> > We've been batting INFO around for many, many years now (like, 10 --
> > it predates the SIP working group). A couple of meetings back, we
> > agreed that we would discuss use cases for INFO packages, and if we
> > didn't find any consensus there, then we would go ahead and  
> publish an
> > "INFO Considered Harmful" RFC.
> >
> > We tried having this discussion at the last IETF, but that just  
> didn't
> > work out.
> >
> > Do we try again, or just give up and publish "INFO Considered
> > Harmful"?
> >
> > Personally, I don't care anymore -- I just want to drive a stake  
> into
> > the heart of this undead-thing, cut off its head, stuff the mouth  
> with
> > holy wafers, and bury it at a crossroads somewhere.
> >
> > But this WG only slays by consensus, so what do you want?
>
> Several people (Jonathan, Mary, Christer, Paul) have responded to the
> list  with various things that added up to "Publish INFO packages AND
> document the extension models for SIP". So far, nobody has suggested
> otherwise.
>
> Does anybody want to do anything else? Speak now or forever visualize
> whirled peas.
>
> --
> Dean
>
>

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


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Thread-Topic: [Sip] INFO and what to do about it?
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From: "DOLLY, MARTIN C, ATTLABS" <mdolly@att.com>
To: <eburger@standardstrack.com>, <SIP@ietf.org>
Subject: Re: [Sip] INFO and what to do about it?
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Deployments and support of INFO is a reality, unlike some of the other
(@#$ in SIP.

So either you believe it is real, or you hide your head under the covers
and pretend that there are no ghosts in the closet. 

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
eburger@standardstrack.com
Sent: Saturday, June 21, 2008 3:11 PM
To: SIP@ietf.org
Subject: Re: [Sip] INFO and what to do about it?

If people care, I'll work, but if nobody cares, I won't.
Sent from my Verizon Wireless BlackBerry

-----Original Message-----
From: Cullen Jennings <fluffy@cisco.com>

Date: Thu, 19 Jun 2008 17:49:00 
To:Dean Willis <dean.willis@softarmor.com>
Cc:sip@ietf.org, Jon Peterson <jon.peterson@neustar.biz>
Subject: Re: [Sip] INFO and what to do about it?



Personally, I'd like to see some coherent discussion around it in a WG  
meeting then try to get to consensus on it. The totally number of  
people bothering to follow the conversation at this point is extremely  
low.

On Jun 18, 2008, at 10:08 PM, Dean Willis wrote:

> I asked:
>
>
> > We've been batting INFO around for many, many years now (like, 10 --
> > it predates the SIP working group). A couple of meetings back, we
> > agreed that we would discuss use cases for INFO packages, and if we
> > didn't find any consensus there, then we would go ahead and  
> publish an
> > "INFO Considered Harmful" RFC.
> >
> > We tried having this discussion at the last IETF, but that just  
> didn't
> > work out.
> >
> > Do we try again, or just give up and publish "INFO Considered
> > Harmful"?
> >
> > Personally, I don't care anymore -- I just want to drive a stake  
> into
> > the heart of this undead-thing, cut off its head, stuff the mouth  
> with
> > holy wafers, and bury it at a crossroads somewhere.
> >
> > But this WG only slays by consensus, so what do you want?
>
> Several people (Jonathan, Mary, Christer, Paul) have responded to the
> list  with various things that added up to "Publish INFO packages AND
> document the extension models for SIP". So far, nobody has suggested
> otherwise.
>
> Does anybody want to do anything else? Speak now or forever visualize
> whirled peas.
>
> --
> Dean
>
>

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


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Cc: sip@ietf.org, Jon Peterson <jon.peterson@neustar.biz>
Subject: Re: [Sip] INFO and what to do about it?
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Cullen Jennings wrote:
> 
> Personally, I'd like to see some coherent discussion around it in a WG 
> meeting then try to get to consensus on it. The totally number of people 
> bothering to follow the conversation at this point is extremely low.

We tried to do that in the last meeting, but I seem to remember AD 
intervention taking our agenda time away.

So it's do it on the list, or don't do it at all, as far as I can tell.

To be honest, I'd say that we have the people who usually pat attention 
to SIP work paying attention now, and the people who usually don't aren't.

--
dean
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Cc: SIP@ietf.org
Subject: Re: [Sip] INFO and what to do about it?
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I'm with Eric on this. I worked for awhile and the results seemed 
futile, so I'm not overly motivated to start again. But if there is some 
possibility of reaching a conclusion then I'll join in.

Certainly I agree with Martin that there are lots of deployments and 
support for INFO in some form. But most of them are non-standard in one 
way or another. And I think this leaves us in a Catch-22 situation:

- we can just bless unrestricted use of INFO. But that would serve no 
purpose, and wouldn't enhance interoperability.

- or we can formalize rules for negotiating usages of INFO. That will 
provide the potential of enhanced interop. But then all the existing 
deployments will be incompatible with it.

	Paul

DOLLY, MARTIN C, ATTLABS wrote:
> Deployments and support of INFO is a reality, unlike some of the other
> (@#$ in SIP.
> 
> So either you believe it is real, or you hide your head under the covers
> and pretend that there are no ghosts in the closet. 
> 
> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> eburger@standardstrack.com
> Sent: Saturday, June 21, 2008 3:11 PM
> To: SIP@ietf.org
> Subject: Re: [Sip] INFO and what to do about it?
> 
> If people care, I'll work, but if nobody cares, I won't.
> Sent from my Verizon Wireless BlackBerry
> 
> -----Original Message-----
> From: Cullen Jennings <fluffy@cisco.com>
> 
> Date: Thu, 19 Jun 2008 17:49:00 
> To:Dean Willis <dean.willis@softarmor.com>
> Cc:sip@ietf.org, Jon Peterson <jon.peterson@neustar.biz>
> Subject: Re: [Sip] INFO and what to do about it?
> 
> 
> 
> Personally, I'd like to see some coherent discussion around it in a WG  
> meeting then try to get to consensus on it. The totally number of  
> people bothering to follow the conversation at this point is extremely  
> low.
> 
> On Jun 18, 2008, at 10:08 PM, Dean Willis wrote:
> 
>> I asked:
>>
>>
>>> We've been batting INFO around for many, many years now (like, 10 --
>>> it predates the SIP working group). A couple of meetings back, we
>>> agreed that we would discuss use cases for INFO packages, and if we
>>> didn't find any consensus there, then we would go ahead and  
>> publish an
>>> "INFO Considered Harmful" RFC.
>>>
>>> We tried having this discussion at the last IETF, but that just  
>> didn't
>>> work out.
>>>
>>> Do we try again, or just give up and publish "INFO Considered
>>> Harmful"?
>>>
>>> Personally, I don't care anymore -- I just want to drive a stake  
>> into
>>> the heart of this undead-thing, cut off its head, stuff the mouth  
>> with
>>> holy wafers, and bury it at a crossroads somewhere.
>>>
>>> But this WG only slays by consensus, so what do you want?
>> Several people (Jonathan, Mary, Christer, Paul) have responded to the
>> list  with various things that added up to "Publish INFO packages AND
>> document the extension models for SIP". So far, nobody has suggested
>> otherwise.
>>
>> Does anybody want to do anything else? Speak now or forever visualize
>> whirled peas.
>>
>> --
>> Dean
>>
>>
> 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
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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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Paul Kyzivat wrote:
> I'm with Eric on this. I worked for awhile and the results seemed
> futile, so I'm not overly motivated to start again. But if there is some
> possibility of reaching a conclusion then I'll join in.
> 
> Certainly I agree with Martin that there are lots of deployments and
> support for INFO in some form. But most of them are non-standard in one
> way or another. And I think this leaves us in a Catch-22 situation:
> 
> - we can just bless unrestricted use of INFO. But that would serve no
> purpose, and wouldn't enhance interoperability.
> 
> - or we can formalize rules for negotiating usages of INFO. That will
> provide the potential of enhanced interop. But then all the existing
> deployments will be incompatible with it.

That sounds like a vote for defaulting and publishing Eric's original
"Why you shouldn't use INFO for anything" draft.

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


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Probably everybody is busy. Any thoughts on this one.


Best Regards,
Ashish Saxena
877-5570

-----Original Message-----
From: Ashish Saxena (WT01 - Telecom Equipment) 
Sent: Friday, June 20, 2008 10:56 AM
To: christer.holmberg@ericsson.com
Cc: sip@ietf.org
Subject: RE: [Sip] Draft submission: draft-ietf-sip-199-00

Hi Christer,
	From Section 4 (Client Behaviour) part of this draft, I
understood that primary objective of this draft is to give indication to
UAC to release resources, if any. I am little worried about extra
messaging that will be seen because of 199 implementation. Can there be
a way to tell proxies and UAS downstream that UAC has some resources
reserved for this call and would be interested in receiving 199, if
possible?

IMO, UAC (mostly) is in better understanding of the resources that it
has reserved for a particular call. So UAC can selectively add something
in INVITE to tell downstream proxies/ UAS' about its interest. 

IMO, we can have a package for 199 and UAC can use "Supported" header to
indicate entities downstream that it would be interested in getting 199
for this *particular* call. Implicitly it would mean that UAC has
reserved some resources. I understand that we would changing the meaning
of "Supported" header on per call basis :-(. Something else may be?

Comments please.

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


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Hi

Can we use Require:199 to indicate to proxy to send 199? 


Regards
Ranjit

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
ashish.hsaxena@wipro.com
Sent: Monday, June 23, 2008 10:24 AM
To: ashish.hsaxena@wipro.com; christer.holmberg@ericsson.com
Cc: sip@ietf.org
Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00


Probably everybody is busy. Any thoughts on this one.


Best Regards,
Ashish Saxena
877-5570

-----Original Message-----
From: Ashish Saxena (WT01 - Telecom Equipment)
Sent: Friday, June 20, 2008 10:56 AM
To: christer.holmberg@ericsson.com
Cc: sip@ietf.org
Subject: RE: [Sip] Draft submission: draft-ietf-sip-199-00

Hi Christer,
	From Section 4 (Client Behaviour) part of this draft, I
understood that primary objective of this draft is to give indication to
UAC to release resources, if any. I am little worried about extra
messaging that will be seen because of 199 implementation. Can there be
a way to tell proxies and UAS downstream that UAC has some resources
reserved for this call and would be interested in receiving 199, if
possible?

IMO, UAC (mostly) is in better understanding of the resources that it
has reserved for a particular call. So UAC can selectively add something
in INVITE to tell downstream proxies/ UAS' about its interest. 

IMO, we can have a package for 199 and UAC can use "Supported" header to
indicate entities downstream that it would be interested in getting 199
for this *particular* call. Implicitly it would mean that UAC has
reserved some resources. I understand that we would changing the meaning
of "Supported" header on per call basis :-(. Something else may be?

Comments please.

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


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> QUESTION 1 TO SIP WG: Are the use cases sufficiently important to
> proceed with this draft? The document states:

I agree this is a relevant use-case. If there's no other one, I think
we don't have to torture ourselves with too many use-cases, as it
would complicate the solution.
By the way, I like short & simple drafts like this one ;-)


> QUESTION 2 TO SIP WG: Do we have a robust set of requirements for
> proceeding with this work? The document currently lists:

I would add one explicit requirement like: It MUST be possible for a
UA to negotiate a keep-alive mecanism outside a SIP registration.


Cheers,
Xavier
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Thanks Ranjit,
	IMO it will be too restrictive on entities because we do not
want to block the call if the entities do not support this package.

Thanks.

Best Regards,
Ashish Saxena
877-5570

-----Original Message-----
From: Avasarala Ranjit-A20990 [mailto:ranjit@motorola.com] 
Sent: Monday, June 23, 2008 12:27 PM
To: Ashish Saxena (WT01 - Telecom Equipment);
christer.holmberg@ericsson.com
Cc: sip@ietf.org
Subject: RE: [Sip] Draft submission: draft-ietf-sip-199-00

Hi

Can we use Require:199 to indicate to proxy to send 199? 


Regards
Ranjit

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
ashish.hsaxena@wipro.com
Sent: Monday, June 23, 2008 10:24 AM
To: ashish.hsaxena@wipro.com; christer.holmberg@ericsson.com
Cc: sip@ietf.org
Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00


Probably everybody is busy. Any thoughts on this one.


Best Regards,
Ashish Saxena
877-5570

-----Original Message-----
From: Ashish Saxena (WT01 - Telecom Equipment)
Sent: Friday, June 20, 2008 10:56 AM
To: christer.holmberg@ericsson.com
Cc: sip@ietf.org
Subject: RE: [Sip] Draft submission: draft-ietf-sip-199-00

Hi Christer,
	From Section 4 (Client Behaviour) part of this draft, I
understood that primary objective of this draft is to give indication to
UAC to release resources, if any. I am little worried about extra
messaging that will be seen because of 199 implementation. Can there be
a way to tell proxies and UAS downstream that UAC has some resources
reserved for this call and would be interested in receiving 199, if
possible?

IMO, UAC (mostly) is in better understanding of the resources that it
has reserved for a particular call. So UAC can selectively add something
in INVITE to tell downstream proxies/ UAS' about its interest. 

IMO, we can have a package for 199 and UAC can use "Supported" header to
indicate entities downstream that it would be interested in getting 199
for this *particular* call. Implicitly it would mean that UAC has
reserved some resources. I understand that we would changing the meaning
of "Supported" header on per call basis :-(. Something else may be?

Comments please.

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


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From: "Xavier Marjou" <xavier.marjou@orange-ftgroup.com>
To: "Dean Willis" <dean.willis@softarmor.com>
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Subject: Re: [Sip] INFO and what to do about it?
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My opinion is that INFO is used a lot in SIP whatever people like it
or not. It should thus be accepted, as well as corrected in its
details if some minor aspects need to be enhanced.

The work done in draft-kaplan-sip-info-events goes this direction and
should thus become a WG item.

Cheers,
Xavier

On Wed, Jun 18, 2008 at 12:27 AM, Dean Willis <dean.willis@softarmor.com> wrote:
>
> We've been batting INFO around for many, many years now (like, 10 --
> it predates the SIP working group). A couple of meetings back, we
> agreed that we would discuss use cases for INFO packages, and if we
> didn't find any consensus there, then we would go ahead and publish an
> "INFO Considered Harmful" RFC.
>
> We tried having this discussion at the last IETF, but that just didn't
> work out.
>
> Do we try again, or just give up and publish "INFO Considered Harmful"?
>
> Personally, I don't care anymore -- I just want to drive a stake into
> the heart of this undead-thing, cut off its head, stuff the mouth with
> holy wafers, and bury it at a crossroads somewhere.
>
> But this WG only slays by consensus, so what do you want?
>
> --
> 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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Hi, 

>Can we use Require:199 to indicate to proxy to send 199? 

No. The Require header is only applicaple to a UAS.

You can use Proxy-Require for proxies, but the problem is that you would
require this from EVERY proxy in the path.

Regards,

Christer



-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
ashish.hsaxena@wipro.com
Sent: Monday, June 23, 2008 10:24 AM
To: ashish.hsaxena@wipro.com; christer.holmberg@ericsson.com
Cc: sip@ietf.org
Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00


Probably everybody is busy. Any thoughts on this one.


Best Regards,
Ashish Saxena
877-5570

-----Original Message-----
From: Ashish Saxena (WT01 - Telecom Equipment)
Sent: Friday, June 20, 2008 10:56 AM
To: christer.holmberg@ericsson.com
Cc: sip@ietf.org
Subject: RE: [Sip] Draft submission: draft-ietf-sip-199-00

Hi Christer,
	From Section 4 (Client Behaviour) part of this draft, I
understood that primary objective of this draft is to give indication to
UAC to release resources, if any. I am little worried about extra
messaging that will be seen because of 199 implementation. Can there be
a way to tell proxies and UAS downstream that UAC has some resources
reserved for this call and would be interested in receiving 199, if
possible?

IMO, UAC (mostly) is in better understanding of the resources that it
has reserved for a particular call. So UAC can selectively add something
in INVITE to tell downstream proxies/ UAS' about its interest. 

IMO, we can have a package for 199 and UAC can use "Supported" header to
indicate entities downstream that it would be interested in getting 199
for this *particular* call. Implicitly it would mean that UAC has
reserved some resources. I understand that we would changing the meaning
of "Supported" header on per call basis :-(. Something else may be?

Comments please.

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

>From Section 4 (Client Behaviour) part of this draft, I understood that
primary objective of this draft is to give 
>indication to UAC to release resources, if any. I am little worried
about extra messaging that will be seen because of 
>199 implementation. Can there be a way to tell proxies and UAS
downstream that UAC has some resources reserved for this 
>call and would be interested in receiving 199, if possible?
>
>IMO, UAC (mostly) is in better understanding of the resources that it
has reserved for a particular call. So UAC can 
>selectively add something in INVITE to tell downstream proxies/ UAS'
about its interest. 
>
>IMO, we can have a package for 199 and UAC can use "Supported" header
to indicate entities downstream that it would be 
>interested in getting 199 for this *particular* call. Implicitly it
would mean that UAC has reserved some resources. I 
>understand that we would changing the meaning of "Supported" header on
per call basis :-(. Something else may be?

I am not sure you would "break" anything by using Supported "per call".
As far as I know, it is nowhere said that a UAC must included the same
option-tags in the Supported header for every call it makes...

It is true that the primary objective is to give the indication to the
UAC, but in some environments there could also be intermediates (P-CSCF
controlling PCC in IMS for example) that are interested in this
indication.

Regards,

Christer

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to, 2008-06-19 kello 11:30 +0200, ext DRAGE, Keith (Keith) kirjoitti:
> QUESTION 1 TO SIP WG: Are the use cases sufficiently important to
> proceed with this draft? The document states:
> 
>    Chapter 3.5 of draft-ietf-sip-outbound-13 [I-D.ietf-sip-outbound]
>    defines two keep-alive techniques.  Even though the keep-alive
>    techniques are separated from the Outbound mechanism
>    [I-D.ietf-sip-outbound], it is currently not possible to indicate
>    support of the keep-alive techniques without also indicating support
>    for the Outbound mechanism.
> 
>    The Outbound mechanism is enabled during the UA registration phase.
>    However, there are use-cases where the UA does not register itself,
>    but still needs to be able to make calls and maintain NAT bindings
>    open during the duration of that call.  A typical example is
>    emergency calls.  There are also cases where entities do not support
>    the Outbound mechanism, but still want to be able to indicate support
>    and use the keep-alive techniques defined in [I-D.ietf-sip-outbound].
> 
> At first sight this is not the most inspiring declaration of the need
> for the document. Please respond indicating whether you consider this a
> useful draft, and propose text that you think would be useful in this
> section. Conversely, if you think this draft is not useful and the WG
> has other more important things to work on first, please also respond.

In theory, there is a small, but valid, reason for the draft to exists.
The draft could serve as a way to deploy outbound-aware proxies in front
of outbound-unaware registrars, using UDP transport. However, arguably
UDP is deployed and works through NAT today, which is why I find it
unlikely that anyone would deploy outbound keepalives in this way.

> QUESTION 2 TO SIP WG: Do we have a robust set of requirements for
> proceeding with this work? The document currently lists:
> 
>    REQ 1: It MUST be possible for a UA to indicate support of the keep-
>    alive techniques defined [I-D.ietf-sip-outbound] if the UA supports
>    only the keep-alive part of [I-D.ietf-sip-outbound].
> 
>    REQ 2: It MUST be possible for an edge proxy to indicate support of
>    the keep-alive techniques defined [I-D.ietf-sip-outbound] if the edge
>    poxy supports only the keep-alive part of [I-D.ietf-sip-outbound].
> 
> It would be desirable to agree these at the outset, and not revisit them
> if we continue with the work. So if you require clarification,
> modification, or addition to these two requirements, then please also
> response with your questions and proposals.

These requirements would make sense, but iff there was an actual need to
have this ability in the first place, as I described above. I don't
think such a need exists in practice.

(Note, this upgrade path only exists for UDP; for TCP registrars need to
be outbound-aware.)

Cheers,
Aki

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Le Thursday 19 June 2008 12:30:30 ext DRAGE, Keith (Keith), vous avez =E9cr=
it=A0:
> QUESTION 1 TO SIP WG: Are the use cases sufficiently important to
> proceed with this draft? The document states:
(...)
>    The Outbound mechanism is enabled during the UA registration phase.
>    However, there are use-cases where the UA does not register itself,
>    but still needs to be able to make calls and maintain NAT bindings
>    open during the duration of that call.

>    A typical example is emergency calls.

I don't get this one. You don't need to send keep-alive packets to make an =

outbound call (be it urgent or not).

>    There are also cases where entities do not support =

>    the Outbound mechanism, but still want to be able to indicate support
>    and use the keep-alive techniques defined in [I-D.ietf-sip-outbound].

I don't get this either. If you don't support Outbound, why not use plain S=
IP =

OPTIONS, which works _today_?

-- =

R=E9mi Denis-Courmont
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Hi, 

>>QUESTION 1 TO SIP WG: Are the use cases sufficiently important to 
>>proceed with this draft? The document states:
>>(...)
>>    The Outbound mechanism is enabled during the UA registration phase.
>>    However, there are use-cases where the UA does not register itself,
>>    but still needs to be able to make calls and maintain NAT bindings
>>    open during the duration of that call.
>>
>>    A typical example is emergency calls.
>
>I don't get this one. You don't need to send keep-alive packets to make an outbound call (be it urgent or not).

I am not sure that is what I am saying. I am saying that you in these use-cases can't use outbound, with keep-alives, even if you support it.

>>    There are also cases where entities do not support 
>>    the Outbound mechanism, but still want to be able to indicate support
>>    and use the keep-alive techniques defined in [I-D.ietf-sip-outbound].
>
>I don't get this either. If you don't support Outbound, why not use plain SIP OPTIONS, which works _today_?

An edge proxy does not know whether the UA is going to send OPTIONS, so it may use the short registration rerefresh mechanism. One of the purposes of the draft is for UAs to tell the edge proxy that it is willing to send keep-alives, so that the edge proxy doesn't need to use registration type of mechanisms.

In addition, if OPTIONS is sent in-dialog it will be forwarded by the edge proxy towards the remote user, which is not needed.

Regards,

Christer

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On Jun 21, 2008, at 3:57 PM, Dean Willis wrote:

> Cullen Jennings wrote:
>> Personally, I'd like to see some coherent discussion around it in a  
>> WG meeting then try to get to consensus on it. The totally number  
>> of people bothering to follow the conversation at this point is  
>> extremely low.
>
> We tried to do that in the last meeting, but I seem to remember AD  
> intervention taking our agenda time away.

Dean, a room full a 100+ people in the sipping meeting took a hum and  
decided much to my surprise to not add an evening meeting. I realize  
we wish there was more time but there is not and it is unlikely there  
will be more time in Dublin. The best we can do is carefully pick what  
topics we do choose to use the time we do have on. I will note there  
was not enough interest in this to get an interim meeting to happen.    
I would not exactly describe this as AD intervention.

>
>
> So it's do it on the list, or don't do it at all, as far as I can  
> tell.
>
> To be honest, I'd say that we have the people who usually pat  
> attention to SIP work paying attention now, and the people who  
> usually don't aren't.
>
> --
> dean

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Cullen Jennings <fluffy@cisco.com>, Dean Willis <dean.willis@softarmor.com>
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> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Cullen Jennings
> Sent: Monday, June 23, 2008 10:41 AM
>
> Dean, a room full a 100+ people in the sipping meeting took a hum and
> decided much to my surprise to not add an evening meeting.

I wasn't there, but my impression from talking with others was that there was concern over having a late-night meeting given some WG's were going to be meeting early the following morning.


> I realize
> we wish there was more time but there is not and it is unlikely there
> will be more time in Dublin. The best we can do is carefully pick what
> topics we do choose to use the time we do have on. I will note there
> was not enough interest in this to get an interim meeting to happen.
> I would not exactly describe this as AD intervention.

I doubt any single topic in the SIP/SIPPING/etc. working groups would by itself be sufficient to warrant an interim meeting, so that's hardly a fair litmus test.

Personally I'd prefer an interim meeting specifically to get most of the current mostly-done WG items out of the way and off of the next full-meeting agenda, and just try to get consensus hums for the outcome of the interim at the full-meeting since they're not likely to be contentious; but my guess is you can't do that at an interim?

-hadriel
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Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
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ashish.hsaxena@wipro.com wrote:
> Probably everybody is busy. Any thoughts on this one.

If the call has been forked, and provisional responses with multiple 
to-tags have been sent back to the UAC, then it is certain that the UAC 
has *some* resources tied up with each early dialog. The 199 will allow 
some of that to be released.

Are you suggesting that the UAC indicate if it considers those resources 
to be sufficient to justify processing an extra message? Or maybe you 
are just concerned with the likelihood that the UAC doesn't support 199, 
so that the message will just be ignored in any case?

I could go either way on this one.

	Paul

> Best Regards,
> Ashish Saxena
> 877-5570
> 
> -----Original Message-----
> From: Ashish Saxena (WT01 - Telecom Equipment) 
> Sent: Friday, June 20, 2008 10:56 AM
> To: christer.holmberg@ericsson.com
> Cc: sip@ietf.org
> Subject: RE: [Sip] Draft submission: draft-ietf-sip-199-00
> 
> Hi Christer,
> 	From Section 4 (Client Behaviour) part of this draft, I
> understood that primary objective of this draft is to give indication to
> UAC to release resources, if any. I am little worried about extra
> messaging that will be seen because of 199 implementation. Can there be
> a way to tell proxies and UAS downstream that UAC has some resources
> reserved for this call and would be interested in receiving 199, if
> possible?
> 
> IMO, UAC (mostly) is in better understanding of the resources that it
> has reserved for a particular call. So UAC can selectively add something
> in INVITE to tell downstream proxies/ UAS' about its interest. 
> 
> IMO, we can have a package for 199 and UAC can use "Supported" header to
> indicate entities downstream that it would be interested in getting 199
> for this *particular* call. Implicitly it would mean that UAC has
> reserved some resources. I understand that we would changing the meaning
> of "Supported" header on per call basis :-(. Something else may be?
> 
> Comments please.
> 
> Best Regards,
> Ashish Saxena
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
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Right - Require is inappropriate. Supported is certainly possible. I 
think it would simply mean that the UAC understands the significance of 
199 and so is prepared to take advantage of it if received. I see no 
reason why you would change it on a call by call basis. The resources 
aren't committed at the time of the INVITE - rather at the time the 
provisional response is received, so doing on a per-call basis wouldn't 
make much sense.

	Paul

Christer Holmberg wrote:
> Hi, 
> 
>> Can we use Require:199 to indicate to proxy to send 199? 
> 
> No. The Require header is only applicaple to a UAS.
> 
> You can use Proxy-Require for proxies, but the problem is that you would
> require this from EVERY proxy in the path.
> 
> Regards,
> 
> Christer
> 
> 
> 
> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> ashish.hsaxena@wipro.com
> Sent: Monday, June 23, 2008 10:24 AM
> To: ashish.hsaxena@wipro.com; christer.holmberg@ericsson.com
> Cc: sip@ietf.org
> Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
> 
> 
> Probably everybody is busy. Any thoughts on this one.
> 
> 
> Best Regards,
> Ashish Saxena
> 877-5570
> 
> -----Original Message-----
> From: Ashish Saxena (WT01 - Telecom Equipment)
> Sent: Friday, June 20, 2008 10:56 AM
> To: christer.holmberg@ericsson.com
> Cc: sip@ietf.org
> Subject: RE: [Sip] Draft submission: draft-ietf-sip-199-00
> 
> Hi Christer,
> 	From Section 4 (Client Behaviour) part of this draft, I
> understood that primary objective of this draft is to give indication to
> UAC to release resources, if any. I am little worried about extra
> messaging that will be seen because of 199 implementation. Can there be
> a way to tell proxies and UAS downstream that UAC has some resources
> reserved for this call and would be interested in receiving 199, if
> possible?
> 
> IMO, UAC (mostly) is in better understanding of the resources that it
> has reserved for a particular call. So UAC can selectively add something
> in INVITE to tell downstream proxies/ UAS' about its interest. 
> 
> IMO, we can have a package for 199 and UAC can use "Supported" header to
> indicate entities downstream that it would be interested in getting 199
> for this *particular* call. Implicitly it would mean that UAC has
> reserved some resources. I understand that we would changing the meaning
> of "Supported" header on per call basis :-(. Something else may be?
> 
> Comments please.
> 
> Best Regards,
> Ashish Saxena
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use
> sip-implementors@cs.columbia.edu for questions on current sip Use
> sipping@ietf.org for new developments on the application of sip
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
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On Jun 23, 2008, at 9:41 AM, Cullen Jennings wrote:

>
> On Jun 21, 2008, at 3:57 PM, Dean Willis wrote:
>
>> Cullen Jennings wrote:
>>> Personally, I'd like to see some coherent discussion around it in  
>>> a WG meeting then try to get to consensus on it. The totally  
>>> number of people bothering to follow the conversation at this  
>>> point is extremely low.
>>
>> We tried to do that in the last meeting, but I seem to remember AD  
>> intervention taking our agenda time away.
>
> Dean, a room full a 100+ people in the sipping meeting took a hum  
> and decided much to my surprise to not add an evening meeting. I  
> realize we wish there was more time but there is not and it is  
> unlikely there will be more time in Dublin. The best we can do is  
> carefully pick what topics we do choose to use the time we do have  
> on. I will note there was not enough interest in this to get an  
> interim meeting to happen.   I would not exactly describe this as AD  
> intervention.

No, I was talking about the 4 hour agenda request that got cut to 2  
hours along with the instruction to "focus only on chartered work".

INFO is interesting, but it's not on our milestone list.

--
Dean
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From sip-bounces@ietf.org  Mon Jun 23 09:18:49 2008
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FYI,  we actually had the meeting targetted as a SIPPING WG adhoc:
https://www3.ietf.org/proceedings/08mar/slides/sipping-0/sipping-0_files
/frame.htm  (Chart 16)

Personally, I was in favor of an interim to discuss that topic as well
as other critical WG items. In the past, interims have really helped in
progressing key WG items.  As I understand, one roadblock for some folks
is travel approval (my proposal had been to hold the interim in Dallas
before or after the GEOPRIV interim).  It might be useful to plan
earlier for interims to allow folks to consider this in travel budgets
sooner (i.e., some of us do these quarterly and having it in the plan
earlier allows it to be approved with all other travel and can be
considered/prioritized with other travel requests, whereas if it comes
later, it requires special approval and is often challenged given
today's business environment). 

I will note that Keith did try to schedule a teleconference on this
topic, however, the quorom of key people were not available.  However,
in general, I do think regular teleconferences for the WGs with authors
to progress key topics would be useful. Recognizing that you can't get
everyone all the time, rotating time slots might help.

I don't think it's apathy or lack of interest in topics as much as many
folks I've communicated with have increased day job workloads, so the
amount of time focused on IETF has dramatically increased for some
folks. 

Mary. 

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Dean Willis
Sent: Monday, June 23, 2008 11:02 AM
To: Cullen Jennings
Cc: sip@ietf.org; Jon Peterson
Subject: Re: [Sip] INFO and what to do about it?


On Jun 23, 2008, at 9:41 AM, Cullen Jennings wrote:

>
> On Jun 21, 2008, at 3:57 PM, Dean Willis wrote:
>
>> Cullen Jennings wrote:
>>> Personally, I'd like to see some coherent discussion around it in a 
>>> WG meeting then try to get to consensus on it. The totally number of

>>> people bothering to follow the conversation at this point is 
>>> extremely low.
>>
>> We tried to do that in the last meeting, but I seem to remember AD 
>> intervention taking our agenda time away.
>
> Dean, a room full a 100+ people in the sipping meeting took a hum and 
> decided much to my surprise to not add an evening meeting. I realize 
> we wish there was more time but there is not and it is unlikely there 
> will be more time in Dublin. The best we can do is carefully pick what

> topics we do choose to use the time we do have on. I will note there 
> was not enough interest in this to get an
> interim meeting to happen.   I would not exactly describe this as AD  
> intervention.

No, I was talking about the 4 hour agenda request that got cut to 2
hours along with the instruction to "focus only on chartered work".

INFO is interesting, but it's not on our milestone list.

--
Dean
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol Use
sip-implementors@cs.columbia.edu for questions on current sip Use
sipping@ietf.org for new developments on the application of sip
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From sip-bounces@ietf.org  Mon Jun 23 09:20:55 2008
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> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Dean Willis
>
> The problem that I'm trying to solve is making the task space
> manageable.
> The first thing is to distribute the chair load.

Obviously splitting/spawning WG's will do that, but I thought there was an AD resource contention issue - if we had more WG meeting slots then neither of our AD's could be in some meeting or other.  Is that not one of the issues?  It was the response I got when I and others suggested SIP Security topics (Identity/Privacy/SIPS/Certs/etc.) should be split into a separate WG a while back.

There was also some concern over not having any more RAI time slots anyway, but I think that's a red herring and overlapping some RAI WG meeting times is reasonable for certain topics.


> I think developing a SIP extension should be a Big Thing. If it's
> important enough to develop a SIP extension for, it is probably
> important enough to charter a working group to develop that extension.
> It's a Really Bad Idea to have a standing working group that sits
> around and develops SIP extensions. All that gets us is a whole lot of
> extensions. It doesn't get us any closer to a draft standard.

I agree with you in principle, but I doubt it's possible in practice.  There's some fine line there, because if you don't address an issue/need multiple people have, then they just go solve it in a proprietary fashion independently and interop still sucks.  One problem is we don't have a single cohesive "group" view and use of SIP, so we're left with the shotgun extensions approach, seeing which ones will stick.  I can't recall another successful protocol this complicated that didn't have one or two dominant vendors, which is a huge disadvantage in some ways.  It's both a blessing and a curse.

-hadriel


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On Jun 23, 2008, at 11:20 AM, Hadriel Kaplan wrote:

>
>> -----Original Message-----
>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
>> Dean Willis
>>
>> The problem that I'm trying to solve is making the task space
>> manageable.
>> The first thing is to distribute the chair load.
>
> Obviously splitting/spawning WG's will do that, but I thought there  
> was an AD resource contention issue - if we had more WG meeting  
> slots then neither of our AD's could be in some meeting or other.   
> Is that not one of the issues?  It was the response I got when I and  
> others suggested SIP Security topics (Identity/Privacy/SIPS/Certs/ 
> etc.) should be split into a separate WG a while back.

> There was also some concern over not having any more RAI time slots  
> anyway, but I think that's a red herring and overlapping some RAI WG  
> meeting times is reasonable for certain topics.
>


We can have four new working groups that meet for an hour each, or we  
can have one existing WG that meets for four hours. Same meeting time,  
same AD conflicts, but four times as many chairs to do the management  
work.

What we can't have is four new working groups that meet for four hours  
each. This means scope creep (and scope inflation) must be rigorously  
controlled. Hence, each charter must be tightly constrained, and each  
WG narrowly focused.

The fact is that our working groups are so big now that the chairs  
can't devote enough time to thoroughly manage individual subjects.  So  
the chairs aren't offloading enough of the work from the ADs. Doubling  
the number of chairs might not cut the ADs load in half, but it could  
certainly reduce it, although this is true if and only if the focus of  
each WG (and chair) is precise. I can't stay on top of the status of  
38 work items that go on for many years. I could probably manage four  
work items that last for no more than two years.

>
>> I think developing a SIP extension should be a Big Thing. If it's
>> important enough to develop a SIP extension for, it is probably
>> important enough to charter a working group to develop that  
>> extension.
>> It's a Really Bad Idea to have a standing working group that sits
>> around and develops SIP extensions. All that gets us is a whole lot  
>> of
>> extensions. It doesn't get us any closer to a draft standard.
>
> I agree with you in principle, but I doubt it's possible in  
> practice.  There's some fine line there, because if you don't  
> address an issue/need multiple people have, then they just go solve  
> it in a proprietary fashion independently and interop still sucks.   
> One problem is we don't have a single cohesive "group" view and use  
> of SIP, so we're left with the shotgun extensions approach, seeing  
> which ones will stick.  I can't recall another successful protocol  
> this complicated that didn't have one or two dominant vendors, which  
> is a huge disadvantage in some ways.  It's both a blessing and a  
> curse.

I expect our "group mind" will continue to exist and evolve. But I'd  
like t have our "Do we need this sort of extension" discussions done  
as BOFs, with full cognizance of the fact that committing resources to  
a new extension is a non-trivial thing. It's far too easy for an  
eternal working group to just keep giving itself new tasks and never  
actually finish.

--
Dean

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From sip-bounces@ietf.org  Mon Jun 23 10:05:48 2008
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	Internet Architecture Board <iab@iab.org>,
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Subject: [Sip] Protocol Action: 'Subscriptions to Request-Contained Resource
 Lists in the Session Initiation Protocol (SIP)' to Proposed Standard
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The IESG has approved the following document:

- 'Subscriptions to Request-Contained Resource Lists in the Session 
   Initiation Protocol (SIP) '
   <draft-ietf-sip-uri-list-subscribe-02.txt> as a Proposed Standard

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

The IESG contact persons are Cullen Jennings and Jon Peterson.

A URL of this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-uri-list-subscribe-02.txt

Technical Summary 

This document specifies a way to create subscription to a list of
resources in SIP. This is achieved by including the list of resources in
the body of a SUBSCRIBE. Instead of having a subscriber send a SUBSCRIBE
for each resource individually, the subscriber defines the resource list,
subscribes to it, and gets notifications about changes in the resources'
state using a single SUBSCRIBE dialog. 

Working Group Summary 

The document was originally produced by the SIPPING working group, but
was transferred to the SIP working group due to the need to define a new
option tag, in conformance with RFC 3427. There is consensus in the WG to
publish this document. 

Document Quality 

There is a strong requirement from OMA for a SIP solution in this area. 

Personnel 

Keith Drage is the document shepherd for this document. Cullen Jennings
is the responsible Area Director.

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Subject: [Sip] Protocol Action: 'Conference Establishment Using
 Request-Contained Lists in the Session Initiation Protocol (SIP)' to
 Proposed Standard
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The IESG has approved the following document:

- 'Conference Establishment Using Request-Contained Lists in the Session 
   Initiation Protocol (SIP) '
   <draft-ietf-sip-uri-list-conferencing-02.txt> as a Proposed Standard

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

The IESG contact persons are Cullen Jennings and Jon Peterson.

A URL of this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-uri-list-conferencing-02.txt

Technical Summary 

This document describes how to create a conference using SIP URI-list 
services. In particular, it describes a mechanism that allows a client to
provide a conference server with the initial list of participants using an
INVITE-contained URI-list. 

Working Group Summary 

The document was originally produced by the SIPPING working group, 
but was transferred to the SIP working group due to the need to define 
a new option tag, in conformance with RFC 3427. 

Document Quality 

There is a strong requirement from OMA and 3GPP for a SIP 
solution in this area. 

Personnel 

Keith Drage is the document shepherd for this document. 
Cullen Jennings is the responsible Area Director.

_______________________________________________
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Cc: SIP@ietf.org, "DOLLY, MARTIN C, ATTLABS" <mdolly@att.com>
Subject: Re: [Sip] INFO and what to do about it?
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Paul's two points resonate with me.

Does anyone expect that if we _did_ build a usage framework for INFO,  
that we would somehow take the existing uses and retroactively declare  
them standard and part of that framework without change?

If that were possible, what's the difference between that and just  
documenting them as is and declaring them standard without this  
framework.

So I don't see how building this framework will solve the kind of  
problem that Martin was punctation-charactering about.

As Paul points out, creating the framework might provide better  
interoperability for some _new_ use, but the level of effort someone  
would have to go through to get the usage standardized is not going to  
be easier than standardizing it on its own. So having its not going to  
do the person that wants a nifty new feature any real good.

If we had a truckload of things that would use the framework bursting  
at the seams waiting for the framework, we'd all, I'm betting,
happily work very quickly to create it. I don't see this truckload of  
things.  The existing uses are out there and they're not going to  
change (so far when I've asked folks with features build on INFO if  
they'd change their code to use this framework once it existed,  
they've laughed).

So if we do anything at all, I'd favor finishing the work Paul started.

RjS



On Jun 21, 2008, at 6:15 PM, Paul Kyzivat wrote:

> I'm with Eric on this. I worked for awhile and the results seemed  
> futile, so I'm not overly motivated to start again. But if there is  
> some possibility of reaching a conclusion then I'll join in.
>
> Certainly I agree with Martin that there are lots of deployments and  
> support for INFO in some form. But most of them are non-standard in  
> one way or another. And I think this leaves us in a Catch-22  
> situation:
>
> - we can just bless unrestricted use of INFO. But that would serve  
> no purpose, and wouldn't enhance interoperability.
>
> - or we can formalize rules for negotiating usages of INFO. That  
> will provide the potential of enhanced interop. But then all the  
> existing deployments will be incompatible with it.
>
> 	Paul
>
> DOLLY, MARTIN C, ATTLABS wrote:
>> Deployments and support of INFO is a reality, unlike some of the  
>> other
>> (@#$ in SIP.
>> So either you believe it is real, or you hide your head under the  
>> covers
>> and pretend that there are no ghosts in the closet. -----Original  
>> Message-----
>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
>> eburger@standardstrack.com
>> Sent: Saturday, June 21, 2008 3:11 PM
>> To: SIP@ietf.org
>> Subject: Re: [Sip] INFO and what to do about it?
>> If people care, I'll work, but if nobody cares, I won't.
>> Sent from my Verizon Wireless BlackBerry
>> -----Original Message-----
>> From: Cullen Jennings <fluffy@cisco.com>
>> Date: Thu, 19 Jun 2008 17:49:00 To:Dean Willis <dean.willis@softarmor.com 
>> >
>> Cc:sip@ietf.org, Jon Peterson <jon.peterson@neustar.biz>
>> Subject: Re: [Sip] INFO and what to do about it?
>> Personally, I'd like to see some coherent discussion around it in a  
>> WG  meeting then try to get to consensus on it. The totally number  
>> of  people bothering to follow the conversation at this point is  
>> extremely  low.
>> On Jun 18, 2008, at 10:08 PM, Dean Willis wrote:
>>> I asked:
>>>
>>>
>>>> We've been batting INFO around for many, many years now (like, 10  
>>>> --
>>>> it predates the SIP working group). A couple of meetings back, we
>>>> agreed that we would discuss use cases for INFO packages, and if we
>>>> didn't find any consensus there, then we would go ahead and
>>> publish an
>>>> "INFO Considered Harmful" RFC.
>>>>
>>>> We tried having this discussion at the last IETF, but that just
>>> didn't
>>>> work out.
>>>>
>>>> Do we try again, or just give up and publish "INFO Considered
>>>> Harmful"?
>>>>
>>>> Personally, I don't care anymore -- I just want to drive a stake
>>> into
>>>> the heart of this undead-thing, cut off its head, stuff the mouth
>>> with
>>>> holy wafers, and bury it at a crossroads somewhere.
>>>>
>>>> But this WG only slays by consensus, so what do you want?
>>> Several people (Jonathan, Mary, Christer, Paul) have responded to  
>>> the
>>> list  with various things that added up to "Publish INFO packages  
>>> AND
>>> document the extension models for SIP". So far, nobody has suggested
>>> otherwise.
>>>
>>> Does anybody want to do anything else? Speak now or forever  
>>> visualize
>>> whirled peas.
>>>
>>> --
>>> Dean
>>>
>>>
>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
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>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

_______________________________________________
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Hi Robert,

INFO is out there, true, and I am not sure we can do much about that. I don=
't think the intention is to standardize the current usages.

I think the idea of the framework is for FUTURE INFO usages. Because, I don=
't think we can assume there won't be more usages in the future than we cur=
rently have - even if we would try to "forbid" INFO.

Regards,

Christer



-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Rober=
t Sparks
Sent: 23. kes=E4kuuta 2008 20:28
To: Paul Kyzivat
Cc: SIP@ietf.org; DOLLY, MARTIN C, ATTLABS
Subject: Re: [Sip] INFO and what to do about it?

Paul's two points resonate with me.

Does anyone expect that if we _did_ build a usage framework for INFO, that =
we would somehow take the existing uses and retroactively declare them stan=
dard and part of that framework without change?

If that were possible, what's the difference between that and just document=
ing them as is and declaring them standard without this framework.

So I don't see how building this framework will solve the kind of problem t=
hat Martin was punctation-charactering about.

As Paul points out, creating the framework might provide better interoperab=
ility for some _new_ use, but the level of effort someone would have to go =
through to get the usage standardized is not going to be easier than standa=
rdizing it on its own. So having its not going to do the person that wants =
a nifty new feature any real good.

If we had a truckload of things that would use the framework bursting at th=
e seams waiting for the framework, we'd all, I'm betting, happily work very=
 quickly to create it. I don't see this truckload of things.  The existing =
uses are out there and they're not going to change (so far when I've asked =
folks with features build on INFO if they'd change their code to use this f=
ramework once it existed, they've laughed).

So if we do anything at all, I'd favor finishing the work Paul started.

RjS



On Jun 21, 2008, at 6:15 PM, Paul Kyzivat wrote:

> I'm with Eric on this. I worked for awhile and the results seemed =

> futile, so I'm not overly motivated to start again. But if there is =

> some possibility of reaching a conclusion then I'll join in.
>
> Certainly I agree with Martin that there are lots of deployments and =

> support for INFO in some form. But most of them are non-standard in =

> one way or another. And I think this leaves us in a Catch-22
> situation:
>
> - we can just bless unrestricted use of INFO. But that would serve no =

> purpose, and wouldn't enhance interoperability.
>
> - or we can formalize rules for negotiating usages of INFO. That will =

> provide the potential of enhanced interop. But then all the existing =

> deployments will be incompatible with it.
>
> 	Paul
>
> DOLLY, MARTIN C, ATTLABS wrote:
>> Deployments and support of INFO is a reality, unlike some of the =

>> other (@#$ in SIP.
>> So either you believe it is real, or you hide your head under the =

>> covers and pretend that there are no ghosts in the closet. =

>> -----Original
>> Message-----
>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =

>> eburger@standardstrack.com
>> Sent: Saturday, June 21, 2008 3:11 PM
>> To: SIP@ietf.org
>> Subject: Re: [Sip] INFO and what to do about it?
>> If people care, I'll work, but if nobody cares, I won't.
>> Sent from my Verizon Wireless BlackBerry -----Original Message-----
>> From: Cullen Jennings <fluffy@cisco.com>
>> Date: Thu, 19 Jun 2008 17:49:00 To:Dean Willis =

>> <dean.willis@softarmor.com
>> >
>> Cc:sip@ietf.org, Jon Peterson <jon.peterson@neustar.biz>
>> Subject: Re: [Sip] INFO and what to do about it?
>> Personally, I'd like to see some coherent discussion around it in a =

>> WG  meeting then try to get to consensus on it. The totally number of  =

>> people bothering to follow the conversation at this point is =

>> extremely  low.
>> On Jun 18, 2008, at 10:08 PM, Dean Willis wrote:
>>> I asked:
>>>
>>>
>>>> We've been batting INFO around for many, many years now (like, 10
>>>> --
>>>> it predates the SIP working group). A couple of meetings back, we =

>>>> agreed that we would discuss use cases for INFO packages, and if we =

>>>> didn't find any consensus there, then we would go ahead and
>>> publish an
>>>> "INFO Considered Harmful" RFC.
>>>>
>>>> We tried having this discussion at the last IETF, but that just
>>> didn't
>>>> work out.
>>>>
>>>> Do we try again, or just give up and publish "INFO Considered =

>>>> Harmful"?
>>>>
>>>> Personally, I don't care anymore -- I just want to drive a stake
>>> into
>>>> the heart of this undead-thing, cut off its head, stuff the mouth
>>> with
>>>> holy wafers, and bury it at a crossroads somewhere.
>>>>
>>>> But this WG only slays by consensus, so what do you want?
>>> Several people (Jonathan, Mary, Christer, Paul) have responded to =

>>> the list  with various things that added up to "Publish INFO =

>>> packages AND document the extension models for SIP". So far, nobody =

>>> has suggested otherwise.
>>>
>>> Does anybody want to do anything else? Speak now or forever =

>>> visualize whirled peas.
>>>
>>> --
>>> Dean
>>>
>>>
>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use =

>> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use =

>> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use =

>> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

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


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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Robert Sparks <rjsparks@nostrum.com>, Paul Kyzivat <pkyzivat@cisco.com>
Date: Mon, 23 Jun 2008 13:40:00 -0400
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What was "the work Paul started"?  You mean Eric's draft?

-hadriel

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Robert Sparks
> Sent: Monday, June 23, 2008 1:28 PM
> To: Paul Kyzivat
> Cc: SIP@ietf.org; DOLLY, MARTIN C, ATTLABS
> Subject: Re: [Sip] INFO and what to do about it?
>
> Paul's two points resonate with me.
>
> Does anyone expect that if we _did_ build a usage framework for INFO,
> that we would somehow take the existing uses and retroactively declare
> them standard and part of that framework without change?
>
> If that were possible, what's the difference between that and just
> documenting them as is and declaring them standard without this
> framework.
>
> So I don't see how building this framework will solve the kind of
> problem that Martin was punctation-charactering about.
>
> As Paul points out, creating the framework might provide better
> interoperability for some _new_ use, but the level of effort someone
> would have to go through to get the usage standardized is not going to
> be easier than standardizing it on its own. So having its not going to
> do the person that wants a nifty new feature any real good.
>
> If we had a truckload of things that would use the framework bursting
> at the seams waiting for the framework, we'd all, I'm betting,
> happily work very quickly to create it. I don't see this truckload of
> things.  The existing uses are out there and they're not going to
> change (so far when I've asked folks with features build on INFO if
> they'd change their code to use this framework once it existed,
> they've laughed).
>
> So if we do anything at all, I'd favor finishing the work Paul started.
>
> RjS
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


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Cc: SIP@ietf.org, "DOLLY, MARTIN C, ATTLABS" <mdolly@att.com>
Subject: Re: [Sip] INFO and what to do about it?
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How about this?

Define the new INFO usage framework, for new uses of INFO, along the 
lines proposed by Hadriel and others. As part of that, establish a 
registry or registries.

Incorporate into that a grandfather clause. If your usage of INFO 
predated some specified date then it will be permitted to be registered, 
but not standardized. Presumably the registration would have to be based 
on Content-Type, since AFAIK that is all that really works for existing 
usages. The registry for these grandfathered usages would have to point 
to *some* specification of the usage. It could be be an individual 
informational RFC, or a document by some other SDO. (What else would 
make sense?)

Existing *standard* usages of INFO (the few there are) would also fall 
under this grandfather clause. In that case there is an RFC to refer to.

*New* usages, after the specified date, would be ineligible for 
registration under the grandfather clause. They would be required to use 
the new formal usage mechanism.

(Damn, I'm allowing myself to be sucked into this again.)

	Thanks,
	Paul

Robert Sparks wrote:
> Paul's two points resonate with me.
> 
> Does anyone expect that if we _did_ build a usage framework for INFO, 
> that we would somehow take the existing uses and retroactively declare 
> them standard and part of that framework without change?
> 
> If that were possible, what's the difference between that and just 
> documenting them as is and declaring them standard without this framework.
> 
> So I don't see how building this framework will solve the kind of 
> problem that Martin was punctation-charactering about.
> 
> As Paul points out, creating the framework might provide better 
> interoperability for some _new_ use, but the level of effort someone 
> would have to go through to get the usage standardized is not going to 
> be easier than standardizing it on its own. So having its not going to 
> do the person that wants a nifty new feature any real good.
> 
> If we had a truckload of things that would use the framework bursting at 
> the seams waiting for the framework, we'd all, I'm betting,
> happily work very quickly to create it. I don't see this truckload of 
> things.  The existing uses are out there and they're not going to change 
> (so far when I've asked folks with features build on INFO if they'd 
> change their code to use this framework once it existed, they've laughed).
> 
> So if we do anything at all, I'd favor finishing the work Paul started.
> 
> RjS
> 
> 
> 
> On Jun 21, 2008, at 6:15 PM, Paul Kyzivat wrote:
> 
>> I'm with Eric on this. I worked for awhile and the results seemed 
>> futile, so I'm not overly motivated to start again. But if there is 
>> some possibility of reaching a conclusion then I'll join in.
>>
>> Certainly I agree with Martin that there are lots of deployments and 
>> support for INFO in some form. But most of them are non-standard in 
>> one way or another. And I think this leaves us in a Catch-22 situation:
>>
>> - we can just bless unrestricted use of INFO. But that would serve no 
>> purpose, and wouldn't enhance interoperability.
>>
>> - or we can formalize rules for negotiating usages of INFO. That will 
>> provide the potential of enhanced interop. But then all the existing 
>> deployments will be incompatible with it.
>>
>>     Paul
>>
>> DOLLY, MARTIN C, ATTLABS wrote:
>>> Deployments and support of INFO is a reality, unlike some of the other
>>> (@#$ in SIP.
>>> So either you believe it is real, or you hide your head under the covers
>>> and pretend that there are no ghosts in the closet. -----Original 
>>> Message-----
>>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
>>> eburger@standardstrack.com
>>> Sent: Saturday, June 21, 2008 3:11 PM
>>> To: SIP@ietf.org
>>> Subject: Re: [Sip] INFO and what to do about it?
>>> If people care, I'll work, but if nobody cares, I won't.
>>> Sent from my Verizon Wireless BlackBerry
>>> -----Original Message-----
>>> From: Cullen Jennings <fluffy@cisco.com>
>>> Date: Thu, 19 Jun 2008 17:49:00 To:Dean Willis 
>>> <dean.willis@softarmor.com>
>>> Cc:sip@ietf.org, Jon Peterson <jon.peterson@neustar.biz>
>>> Subject: Re: [Sip] INFO and what to do about it?
>>> Personally, I'd like to see some coherent discussion around it in a 
>>> WG  meeting then try to get to consensus on it. The totally number 
>>> of  people bothering to follow the conversation at this point is 
>>> extremely  low.
>>> On Jun 18, 2008, at 10:08 PM, Dean Willis wrote:
>>>> I asked:
>>>>
>>>>
>>>>> We've been batting INFO around for many, many years now (like, 10 --
>>>>> it predates the SIP working group). A couple of meetings back, we
>>>>> agreed that we would discuss use cases for INFO packages, and if we
>>>>> didn't find any consensus there, then we would go ahead and
>>>> publish an
>>>>> "INFO Considered Harmful" RFC.
>>>>>
>>>>> We tried having this discussion at the last IETF, but that just
>>>> didn't
>>>>> work out.
>>>>>
>>>>> Do we try again, or just give up and publish "INFO Considered
>>>>> Harmful"?
>>>>>
>>>>> Personally, I don't care anymore -- I just want to drive a stake
>>>> into
>>>>> the heart of this undead-thing, cut off its head, stuff the mouth
>>>> with
>>>>> holy wafers, and bury it at a crossroads somewhere.
>>>>>
>>>>> But this WG only slays by consensus, so what do you want?
>>>> Several people (Jonathan, Mary, Christer, Paul) have responded to the
>>>> list  with various things that added up to "Publish INFO packages AND
>>>> document the extension models for SIP". So far, nobody has suggested
>>>> otherwise.
>>>>
>>>> Does anybody want to do anything else? Speak now or forever visualize
>>>> whirled peas.
>>>>
>>>> -- 
>>>> Dean
>>>>
>>>>
>>> _______________________________________________
>>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol
>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>> Use sipping@ietf.org for new developments on the application of sip
>>> _______________________________________________
>>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol
>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>> Use sipping@ietf.org for new developments on the application of sip
>>> _______________________________________________
>>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol
>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>> Use sipping@ietf.org for new developments on the application of sip
>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
> 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
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Cc: "SIP@ietf.org" <SIP@ietf.org>
Subject: Re: [Sip] INFO and what to do about it?
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I forget who the last proposal was from, but it wasn't me.

	Paul

Hadriel Kaplan wrote:
> What was "the work Paul started"?  You mean Eric's draft?
> 
> -hadriel
> 
>> -----Original Message-----
>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
>> Robert Sparks
>> Sent: Monday, June 23, 2008 1:28 PM
>> To: Paul Kyzivat
>> Cc: SIP@ietf.org; DOLLY, MARTIN C, ATTLABS
>> Subject: Re: [Sip] INFO and what to do about it?
>>
>> Paul's two points resonate with me.
>>
>> Does anyone expect that if we _did_ build a usage framework for INFO,
>> that we would somehow take the existing uses and retroactively declare
>> them standard and part of that framework without change?
>>
>> If that were possible, what's the difference between that and just
>> documenting them as is and declaring them standard without this
>> framework.
>>
>> So I don't see how building this framework will solve the kind of
>> problem that Martin was punctation-charactering about.
>>
>> As Paul points out, creating the framework might provide better
>> interoperability for some _new_ use, but the level of effort someone
>> would have to go through to get the usage standardized is not going to
>> be easier than standardizing it on its own. So having its not going to
>> do the person that wants a nifty new feature any real good.
>>
>> If we had a truckload of things that would use the framework bursting
>> at the seams waiting for the framework, we'd all, I'm betting,
>> happily work very quickly to create it. I don't see this truckload of
>> things.  The existing uses are out there and they're not going to
>> change (so far when I've asked folks with features build on INFO if
>> they'd change their code to use this framework once it existed,
>> they've laughed).
>>
>> So if we do anything at all, I'd favor finishing the work Paul started.
>>
>> RjS
> 
_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip


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From: "Mary Barnes" <mary.barnes@nortel.com>
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	"Robert Sparks" <rjsparks@nostrum.com>, "Paul Kyzivat" <pkyzivat@cisco.com>
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Subject: Re: [Sip] INFO and what to do about it?
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I don't believe we could ever forbid INFO. I initially did not think we cou=
ld accomplish anything around INFO, but I believe some of the work that's o=
n the table would be useful for working towards interoperabilty for the INF=
O usages. I would be afraid to ask honestly for the identification of all t=
he different uses of INFO that are out there right now. =


Doing something is better than nothing at this point IMHO and I'm personall=
y really tired of revisiting this issue every couple of years. AND, this wo=
uld help us put a stake in the group on the future usages of INFO (whether =
we ever get rid of the old usages or not), as I believe there are other SDO=
s defining new uses of INFO right now to add to the mix of un-interoperabil=
ity in this area.

My two cents,
Mary. =


-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Chris=
ter Holmberg
Sent: Monday, June 23, 2008 12:34 PM
To: Robert Sparks; Paul Kyzivat
Cc: SIP@ietf.org; DOLLY, MARTIN C, ATTLABS
Subject: Re: [Sip] INFO and what to do about it?


Hi Robert,

INFO is out there, true, and I am not sure we can do much about that. I don=
't think the intention is to standardize the current usages.

I think the idea of the framework is for FUTURE INFO usages. Because, I don=
't think we can assume there won't be more usages in the future than we cur=
rently have - even if we would try to "forbid" INFO.

Regards,

Christer



-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Rober=
t Sparks
Sent: 23. kes=E4kuuta 2008 20:28
To: Paul Kyzivat
Cc: SIP@ietf.org; DOLLY, MARTIN C, ATTLABS
Subject: Re: [Sip] INFO and what to do about it?

Paul's two points resonate with me.

Does anyone expect that if we _did_ build a usage framework for INFO, that =
we would somehow take the existing uses and retroactively declare them stan=
dard and part of that framework without change?

If that were possible, what's the difference between that and just document=
ing them as is and declaring them standard without this framework.

So I don't see how building this framework will solve the kind of problem t=
hat Martin was punctation-charactering about.

As Paul points out, creating the framework might provide better interoperab=
ility for some _new_ use, but the level of effort someone would have to go =
through to get the usage standardized is not going to be easier than standa=
rdizing it on its own. So having its not going to do the person that wants =
a nifty new feature any real good.

If we had a truckload of things that would use the framework bursting at th=
e seams waiting for the framework, we'd all, I'm betting, happily work very=
 quickly to create it. I don't see this truckload of things.  The existing =
uses are out there and they're not going to change (so far when I've asked =
folks with features build on INFO if they'd change their code to use this f=
ramework once it existed, they've laughed).

So if we do anything at all, I'd favor finishing the work Paul started.

RjS



On Jun 21, 2008, at 6:15 PM, Paul Kyzivat wrote:

> I'm with Eric on this. I worked for awhile and the results seemed =

> futile, so I'm not overly motivated to start again. But if there is =

> some possibility of reaching a conclusion then I'll join in.
>
> Certainly I agree with Martin that there are lots of deployments and =

> support for INFO in some form. But most of them are non-standard in =

> one way or another. And I think this leaves us in a Catch-22
> situation:
>
> - we can just bless unrestricted use of INFO. But that would serve no =

> purpose, and wouldn't enhance interoperability.
>
> - or we can formalize rules for negotiating usages of INFO. That will =

> provide the potential of enhanced interop. But then all the existing =

> deployments will be incompatible with it.
>
> 	Paul
>
> DOLLY, MARTIN C, ATTLABS wrote:
>> Deployments and support of INFO is a reality, unlike some of the =

>> other (@#$ in SIP.
>> So either you believe it is real, or you hide your head under the =

>> covers and pretend that there are no ghosts in the closet.
>> -----Original
>> Message-----
>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =

>> eburger@standardstrack.com
>> Sent: Saturday, June 21, 2008 3:11 PM
>> To: SIP@ietf.org
>> Subject: Re: [Sip] INFO and what to do about it?
>> If people care, I'll work, but if nobody cares, I won't.
>> Sent from my Verizon Wireless BlackBerry -----Original Message-----
>> From: Cullen Jennings <fluffy@cisco.com>
>> Date: Thu, 19 Jun 2008 17:49:00 To:Dean Willis =

>> <dean.willis@softarmor.com
>> >
>> Cc:sip@ietf.org, Jon Peterson <jon.peterson@neustar.biz>
>> Subject: Re: [Sip] INFO and what to do about it?
>> Personally, I'd like to see some coherent discussion around it in a =

>> WG  meeting then try to get to consensus on it. The totally number of =

>> people bothering to follow the conversation at this point is =

>> extremely  low.
>> On Jun 18, 2008, at 10:08 PM, Dean Willis wrote:
>>> I asked:
>>>
>>>
>>>> We've been batting INFO around for many, many years now (like, 10
>>>> --
>>>> it predates the SIP working group). A couple of meetings back, we =

>>>> agreed that we would discuss use cases for INFO packages, and if we =

>>>> didn't find any consensus there, then we would go ahead and
>>> publish an
>>>> "INFO Considered Harmful" RFC.
>>>>
>>>> We tried having this discussion at the last IETF, but that just
>>> didn't
>>>> work out.
>>>>
>>>> Do we try again, or just give up and publish "INFO Considered =

>>>> Harmful"?
>>>>
>>>> Personally, I don't care anymore -- I just want to drive a stake
>>> into
>>>> the heart of this undead-thing, cut off its head, stuff the mouth
>>> with
>>>> holy wafers, and bury it at a crossroads somewhere.
>>>>
>>>> But this WG only slays by consensus, so what do you want?
>>> Several people (Jonathan, Mary, Christer, Paul) have responded to =

>>> the list  with various things that added up to "Publish INFO =

>>> packages AND document the extension models for SIP". So far, nobody =

>>> has suggested otherwise.
>>>
>>> Does anybody want to do anything else? Speak now or forever =

>>> visualize whirled peas.
>>>
>>> --
>>> Dean
>>>
>>>
>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use =

>> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use =

>> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use =

>> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

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


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But you did do a lot of the early hard lifting and that's what I was  
nodding towards.

On Jun 23, 2008, at 12:48 PM, Paul Kyzivat wrote:

> I forget who the last proposal was from, but it wasn't me.
>
> 	Paul
>
> Hadriel Kaplan wrote:
>> What was "the work Paul started"?  You mean Eric's draft?
>> -hadriel
>>> -----Original Message-----
>>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf  
>>> Of
>>> Robert Sparks
>>> Sent: Monday, June 23, 2008 1:28 PM
>>> To: Paul Kyzivat
>>> Cc: SIP@ietf.org; DOLLY, MARTIN C, ATTLABS
>>> Subject: Re: [Sip] INFO and what to do about it?
>>>
>>> Paul's two points resonate with me.
>>>
>>> Does anyone expect that if we _did_ build a usage framework for  
>>> INFO,
>>> that we would somehow take the existing uses and retroactively  
>>> declare
>>> them standard and part of that framework without change?
>>>
>>> If that were possible, what's the difference between that and just
>>> documenting them as is and declaring them standard without this
>>> framework.
>>>
>>> So I don't see how building this framework will solve the kind of
>>> problem that Martin was punctation-charactering about.
>>>
>>> As Paul points out, creating the framework might provide better
>>> interoperability for some _new_ use, but the level of effort someone
>>> would have to go through to get the usage standardized is not  
>>> going to
>>> be easier than standardizing it on its own. So having its not  
>>> going to
>>> do the person that wants a nifty new feature any real good.
>>>
>>> If we had a truckload of things that would use the framework  
>>> bursting
>>> at the seams waiting for the framework, we'd all, I'm betting,
>>> happily work very quickly to create it. I don't see this truckload  
>>> of
>>> things.  The existing uses are out there and they're not going to
>>> change (so far when I've asked folks with features build on INFO if
>>> they'd change their code to use this framework once it existed,
>>> they've laughed).
>>>
>>> So if we do anything at all, I'd favor finishing the work Paul  
>>> started.
>>>
>>> RjS

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> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Dean
> Willis
> Sent: Monday, June 23, 2008 12:16 AM
> >
> > Certainly I agree with Martin that there are lots of deployments and
> > support for INFO in some form. But most of them are non-standard in one
> > way or another. And I think this leaves us in a Catch-22 situation:
> >
> > - we can just bless unrestricted use of INFO. But that would serve no
> > purpose, and wouldn't enhance interoperability.

The EXACT same thing could be said for deprecating/outlawing info:
"We can just restrict the use of INFO.  But that would serve no purpose, and wouldn't enhance interoperability."


> > - or we can formalize rules for negotiating usages of INFO. That will
> > provide the potential of enhanced interop. But then all the existing
> > deployments will be incompatible with it.
>
> That sounds like a vote for defaulting and publishing Eric's original
> "Why you shouldn't use INFO for anything" draft.

Ummm... I don't see how that solves the: "But then all existing deployments will be incompatible with it" concern.  All existing deployments would _still_ be incompatible with such a "don't-use info" draft.

Or is the argument: "there would be no way to make this backwards compatible"?  I think it would be backwards compatible.

-hadriel
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On Jun 23, 2008, at 12:48 PM, Paul Kyzivat wrote:

> How about this?
>
> Define the new INFO usage framework, for new uses of INFO, along the  
> lines proposed by Hadriel and others. As part of that, establish a  
> registry or registries.
>
> Incorporate into that a grandfather clause. If your usage of INFO  
> predated some specified date then it will be permitted to be  
> registered, but not standardized. Presumably the registration would  
> have to be based on Content-Type, since AFAIK that is all that  
> really works for existing usages. The registry for these  
> grandfathered usages would have to point to *some* specification of  
> the usage. It could be be an individual informational RFC, or a  
> document by some other SDO. (What else would make sense?)
>
> Existing *standard* usages of INFO (the few there are) would also  
> fall under this grandfather clause. In that case there is an RFC to  
> refer to.
>
> *New* usages, after the specified date, would be ineligible for  
> registration under the grandfather clause. They would be required to  
> use the new formal usage mechanism.
>

That works for me. It's quite rational, given where we find ourselves,  
and fits with the goal of "reducing further damage."

> (Damn, I'm allowing myself to be sucked into this again.)

I'm so sorry ;-).

--
Dean
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Four 1hr meetings does not equal one four hour meeting. With the time it 
takes for a prior group to clear a room and another group to enter, 
achieve quorum, and cover clerical details, you have lost probably 20 
minutes at a minimum. So your four one hour meetings provide 
4*(60-20)=160 minutes of useful time. A four hour meeting has 
4*60-20=220 minutes. You just lost an hour of working time.

	Paul

Dean Willis wrote:
> 
> On Jun 23, 2008, at 11:20 AM, Hadriel Kaplan wrote:
> 
>>
>>> -----Original Message-----
>>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
>>> Dean Willis
>>>
>>> The problem that I'm trying to solve is making the task space
>>> manageable.
>>> The first thing is to distribute the chair load.
>>
>> Obviously splitting/spawning WG's will do that, but I thought there 
>> was an AD resource contention issue - if we had more WG meeting slots 
>> then neither of our AD's could be in some meeting or other.  Is that 
>> not one of the issues?  It was the response I got when I and others 
>> suggested SIP Security topics (Identity/Privacy/SIPS/Certs/etc.) 
>> should be split into a separate WG a while back.
> 
>> There was also some concern over not having any more RAI time slots 
>> anyway, but I think that's a red herring and overlapping some RAI WG 
>> meeting times is reasonable for certain topics.
>>
> 
> 
> We can have four new working groups that meet for an hour each, or we 
> can have one existing WG that meets for four hours. Same meeting time, 
> same AD conflicts, but four times as many chairs to do the management work.
> 
> What we can't have is four new working groups that meet for four hours 
> each. This means scope creep (and scope inflation) must be rigorously 
> controlled. Hence, each charter must be tightly constrained, and each WG 
> narrowly focused.
> 
> The fact is that our working groups are so big now that the chairs can't 
> devote enough time to thoroughly manage individual subjects.  So the 
> chairs aren't offloading enough of the work from the ADs. Doubling the 
> number of chairs might not cut the ADs load in half, but it could 
> certainly reduce it, although this is true if and only if the focus of 
> each WG (and chair) is precise. I can't stay on top of the status of 38 
> work items that go on for many years. I could probably manage four work 
> items that last for no more than two years.
> 
>>
>>> I think developing a SIP extension should be a Big Thing. If it's
>>> important enough to develop a SIP extension for, it is probably
>>> important enough to charter a working group to develop that extension.
>>> It's a Really Bad Idea to have a standing working group that sits
>>> around and develops SIP extensions. All that gets us is a whole lot of
>>> extensions. It doesn't get us any closer to a draft standard.
>>
>> I agree with you in principle, but I doubt it's possible in practice.  
>> There's some fine line there, because if you don't address an 
>> issue/need multiple people have, then they just go solve it in a 
>> proprietary fashion independently and interop still sucks.  One 
>> problem is we don't have a single cohesive "group" view and use of 
>> SIP, so we're left with the shotgun extensions approach, seeing which 
>> ones will stick.  I can't recall another successful protocol this 
>> complicated that didn't have one or two dominant vendors, which is a 
>> huge disadvantage in some ways.  It's both a blessing and a curse.
> 
> I expect our "group mind" will continue to exist and evolve. But I'd 
> like t have our "Do we need this sort of extension" discussions done as 
> BOFs, with full cognizance of the fact that committing resources to a 
> new extension is a non-trivial thing. It's far too easy for an eternal 
> working group to just keep giving itself new tasks and never actually 
> finish.
> 
> -- 
> Dean
> 
> _______________________________________________
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> 
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On Jun 23, 2008, at 1:38 PM, Paul Kyzivat wrote:

> Four 1hr meetings does not equal one four hour meeting. With the  
> time it takes for a prior group to clear a room and another group to  
> enter, achieve quorum, and cover clerical details, you have lost  
> probably 20 minutes at a minimum. So your four one hour meetings  
> provide 4*(60-20)=160 minutes of useful time. A four hour meeting  
> has 4*60-20=220 minutes. You just lost an hour of working time.

Personally, I gained about 4 hours of working time . . .

Seriously, I don't think the overhead is as high as you estimate, but  
it's certainly not "zero".

However:

If the 4 meetings were better run and better focused because they had  
clear scope, comprehensible agendas, and chairs with enough time to do  
the job right, I believe we'd still come out ahead over what we have  
now.

--
Dean

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From sip-bounces@ietf.org  Mon Jun 23 13:11:54 2008
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Subject: [Sip] Draft Proto-Shepherd writeup for draft-ietf-sip-xcapevent-0
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I believe we're about ready to send the XCAP Event Package draft to  
the IESG. I believe I will be serving as the Document Shepherd. As  
shepherd, I get to do the writeup that accompanies the draft onto the  
iESG docket. Here's my working draft of that writeup. If you have  
anything to add, please do do.


  (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?

The Document Shepherd is SIP working group chair Dean Willis. He has
personally reviewed this version of the document and believes it is
ready for forwrding to the 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 document has been thoroughly reviewed in the working
group. Further review was requested and received from the Open Mobile
Alliance's Presence and Availability working group, which is expected
to be a major user of the specification. Specialized event-package
review was provided by Adam Roach, author of the SIP Events
specification RFC 3325.


    (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 shepherd has no concerns that the document requires additional or
broader review.


    (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 shepherd has no specific concerns with this document, and is
unaware of any IPR disclosure related to 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?

There is strong working group consensus behind this document. It is
the evolution of several years work related to use of XCAP for
configuration delivery and for managing subscriber and buddy lists.


    (1.f)  Has anyone threatened an appeal or otherwise indicated  
extreme
           discontent?  If so, please summarise 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.)

The shepherd is not aware of any significant discontent related to
this draft.


    (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?

The shepherd verified that the draft satisfies all ID nits to the best
of his ability. It does have one slightly stale reference to
draft-ietf-simple-xcap-diff-08 which is now in -09, but this does not
impact the 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].

References are appropriately split. The draft does depend on
draft-ietf-simple-xcap-diff and on draft-ietf-sip-subnot-etags, both
of which should be ready for review in approximately the same time
frame. There are no known downrefs.


    (1.i)  Has the Document Shepherd verified that the document IANA
           consideration 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 Shepherd
           conferred with the Responsible Area Director so that the IESG
           can appoint the needed Expert during the IESG Evaluation?

The IANA considerations appear to be correct.


    (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 shepherd verified the BNF and XML. Since the XML was verified with
the author's own popular XML verifier, no errors were really expected.


    (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

The XCAP (XML Confgiration Access Protocol, RFC 4825) protocol
provides for change control on structured data. This specification
provides a SIP Event Paackage (RFC 3325) that can be used to monitor
changes to an XCAP document. Change notifications may include
differential changes expressed using the XCAP-Diff format described in
draft-ietf-simple-xcap-diff. This specification also provides for
initial synchronization between a stored version of an XCAP document
and a remote copy being monitored using the protocol of thsi  
specification.


Working Group Summary

This specification developed out of the larger body of configuration
management work related to SIP (RFC 3261). In particular, it relates
to XCAP (RFC 4825) and the SIP Configuration Framework
(draft-ietf-sipping-config-framework). The content of this
specification was split off from the configuration framework document
about two years ago, and has progressed indepdently.


           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?

Document Quality

There are existing implementations of earlier versions of the
protocol, and a significant number of vendors have committed to
implementation of the specification in the context of the Open Mobile
Alliances's Push-to-Talk Over Cellularspecification. Adam Roach
provided specific guidance and review on the SIP Event Package aspects
of this specification.


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From sip-bounces@ietf.org  Mon Jun 23 13:23:15 2008
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Dean Willis <dean.willis@softarmor.com>, Paul Kyzivat <pkyzivat@cisco.com>
Date: Mon, 23 Jun 2008 16:23:10 -0400
Thread-Topic: [Sip] Toward the Evolution of SIP and Related Working Groups
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[fixed subject title, since I apparently broke it]

How about a compromise: 2 WG's of 2-hour meetings each (since today's SIP WG actually takes 2 of 2-hour meeting slots I think).
One for the general SIP stuff as today, and one for "SIP Security".

The new security group would take such things as:
draft-ietf-sip-dtls-srtp-framework
draft-ietf-sip-eku
draft-ietf-sip-media-security-requirements
draft-ietf-sip-ua-privacy
draft-ietf-sip-domain-certs
draft-ietf-sip-e2m-sec
draft-ietf-sip-saml

And it would be the venue for any new drafts/work related to Identity, Privacy, transport/media security as relates to SIP, etc.  I think this is roughly half the work in the SIP WG.  The people attending these two WG's would probably be nearly identical as today's SIP WG, and they would consume the same amount of meeting time as today's SIP WG, but it would cut your chair work in half.

It *won't* solve your eternal WG issue, but I think that problem is orthogonal and separately debatable.

-hadriel


> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Dean
> Willis
> Sent: Monday, June 23, 2008 3:27 PM
> To: Paul Kyzivat
> Cc: sip@ietf.org
> Subject: Re: [Sip] (no subject)
>
>
> On Jun 23, 2008, at 1:38 PM, Paul Kyzivat wrote:
>
> > Four 1hr meetings does not equal one four hour meeting. With the
> > time it takes for a prior group to clear a room and another group to
> > enter, achieve quorum, and cover clerical details, you have lost
> > probably 20 minutes at a minimum. So your four one hour meetings
> > provide 4*(60-20)=160 minutes of useful time. A four hour meeting
> > has 4*60-20=220 minutes. You just lost an hour of working time.
>
> Personally, I gained about 4 hours of working time . . .
>
> Seriously, I don't think the overhead is as high as you estimate, but
> it's certainly not "zero".
>
> However:
>
> If the 4 meetings were better run and better focused because they had
> clear scope, comprehensible agendas, and chairs with enough time to do
> the job right, I believe we'd still come out ahead over what we have
> now.
>
> --
> Dean
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
_______________________________________________
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Subject: Re: [Sip] Toward the Evolution of SIP and Related Working Groups
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On Jun 23, 2008, at 3:23 PM, Hadriel Kaplan wrote:

> [fixed subject title, since I apparently broke it]
>
> How about a compromise: 2 WG's of 2-hour meetings each (since  
> today's SIP WG actually takes 2 of 2-hour meeting slots I think).
> One for the general SIP stuff as today, and one for "SIP Security".

  Ah yes, two eternal working groups to replace one.

> The new security group would take such things as:
> draft-ietf-sip-dtls-srtp-framework

-- This one is arguably done, depending on what we believe about RFC  
4474 and gateways/b2buas.

>
> draft-ietf-sip-eku

-- EKU is basically done.

>
> draft-ietf-sip-media-security-requirements

-- This one is done

>
> draft-ietf-sip-ua-privacy

-- This one is done,

>
> draft-ietf-sip-domain-certs

-- This one is done

>
> draft-ietf-sip-e2m-sec

-- This one is done

>
> draft-ietf-sip-saml
-- I'm not sure this one is ever getting done.

So I'm not sure there's enough there to justify a WG.

How about an "Identity in SIP" working group that takes on fixing RFC  
4474 for gateways/b2buas and possibly considers identity/role  
assertion using SAML?

--
Dean

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Subject: Re: [Sip] Toward the Evolution of SIP and Related Working Groups
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My two cents:

I like the idea of splitting SIP into smaller working groups which have a
defined deliverable and finite lifespan.  Doing this does not reduce the work
of the individual contributors, but provides a different benefit:  it
demonstrates interest in specific issues and problems (by the attendance and
activity of the more-focused working group), and it allows working group
chairs to be successful.  In a standards organization, 'success' is defined as
finishing something.  Unfortunately with a never-end set of tasks in front of
SIP (and SIPPPING, MMUSIC, AVT, and probably others), there is never a sense
of completion.  Nothing is 'done'.

I know that we have had difficulty getting chairs for various RAI working
groups.  By creating working groups that have a narrower charter, clear
deliverable, and achievable milestones, we can adopt a more normal structure
-- a structure where an individual contributor can reasonably spend 2-3 years
chairing a working group rather than taking on the significant task of
spinning up to understand all of SIP, SIPPING, MMUSIC, or AVT, and having to
resign after 5 (or 9 years) because they are burned out and no longer
interested in chairing.

-d


> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of Dean Willis
> Sent: Monday, June 16, 2008 4:23 PM
> To: sip@ietf.org
> Subject: [Sip] Toward the Evolution of SIP and Related Working Groups
> 
> 
> The SIP working group spun off from the MMUSIC working group when we  
> realized that the SIP work was reasonably separable from the other  
> work going on in MMUSIC.
> 
> The SIP working group first met officially at IETF 46 in November  
> 1999, although we did have an earlier interim meeting.
> 
> This means that the fall 1999 meeting will be our 10th 
> anniversary. In  
> addition to having a big party, I'd also like to plan for this being  
> the last meeting of the SIP working group as it is now defined. Ten  
> years is long enough. It's time we restructured.
> 
> There will probably still be people who want to change, revise,  
> extend, or otherwise manipulate the family of SIP 
> specifications after  
> 2009.
> 
> So what should we do? How should we organize to get those 
> needs met?   
> Rest assured, we'll be having further discussions on this 
> topic in the  
> coming months. But I'd like to start the discussion now.
> 
> By November 2009, I fully expect SIP to have completed the 
> majority of  
> currently chartered items. Between now and then, we'll probably also  
> take on and complete a bit of new work (although I plan to be very  
> resistant to new work). I also expect that there are a few items on  
> our charter now that we're not going to finish. It may well be that  
> some of those items need a different organizational structure to  
> progress than what we have. Or maybe they don't need doing.
> 
> Further, I've come to believe that most working groups should be  
> shorter-lived and tightly constrained by their charters to produce a  
> single concise set of deliverables. There's still a need for 
> a longer- 
> lived thing that maintains architectural oversight. I've come to  
> believe that this is the "area", not the working group.
> 
> Let's start the recap by looking at the current charter of 
> the SIP WG.  
> I've eliminated the WGLC milestones to shorten the list, as 
> each has a  
> following IESG milestone. Note that there's a Feb 2008 milestone for  
> "Revise Charter". You can consider this topic the first 
> installment on  
> that milestone, even though it is a bit late.
> 
> 1) Jul 2007 Location Conveyance with SIP to IESG (PS)
> 
> 2) Aug 2007 Connection reuse mechanism to IESG (PS)
> 
> 3) Sep 2007 Session Policies to IESG as PS
> 
> 4) Sep 2007 Example security flows to IESG (Informational)
> 
> 5) Oct 2007 New resource priority namespaces for DISA to IESG (PS)
> 
> 6) Nov 2007 Requirements for media keying to IESG (Informational)
> 
> 7 )Dec 2007 Extensions to SIP UA Profile Delivery Change 
> Notification  
> Event Package for XCAP to IESG (PS)
> 
> 8) Dec 2007 Roadmap for SIP to IESG (Informational)
> 
> 9) Dec 2007 Using SAML for SIP to IESG (PS)
> 
> 10) Dec 2007 Extension for use in etags in conditional 
> notification to  
> IESG (PS)
> 
> 11) Feb 2008 Identify requirements for test matrix to move SIP to  
> Draft Standard
> 
> 12) Feb 2008 Delivering request-URI and parameters to UAS via 
> proxy to  
> IESG (PS)
> 
> 13) Feb 2008 Revise charter
> 
> 14) Feb 2008 Establishment of secure media sessions using 
> DTLS-SRTP to  
> IESG (PS)
> 
> 15) Apr 2008 MIME body handling in SIP to IESG (PS)
> 
> 16) Apr 2008 Essential corrections to RFC 3261 (1st batch) to 
> IESG (PS)
> 
> 17) Jul 2008 Mechanisms for UA initiated privacy to IESG (BCP)
> 
> 18) Aug 2008 Guidelines for double route recording to IESG (BCP)
> 
> 19) Sep 2008 X.509 Certificates for TLS use in SIP to IESG (PS)
> 
> 20) Sep 2008 X.509 extended key usage for SIP to IESG (PS)
> 
> 21) Nov 2008 Termination of early dialog prior to final response to  
> IESG (PS)
> 
> 
> Looking at this, we have a couple of discrete tasks that I 
> think might  
> make the basis for different narrowly-scoped working groups. We've  
> also got a lot of stuff that we're late on delivering!
> 
> 1) SIP WG (to finish by end of 2009) gets the bulk of these items.
> 
> 2) SIP Draft Standard WG: Takes the milestone of "Identify  
> requirements for test matrix to move SIP to Draft Standard" 
> and all of  
> the follow-on work needed to move SIP from Proposed to Draft on the  
> standards track. There's certainly enough work here for a dedicated  
> working group, with the last milestone being something like "Deliver  
> SIP as a Draft Standards Document to IESG". We may wish to consider  
> whether we really want to go this route or whether we should instead  
> be thinking of a "SIP 3.0" effort. My personal though is that 
> there's  
> just too much cruft in the RFC 3261 family to make Draft, but 
> I could  
> be proven wrong. I'd rather see us put SIP 2.0 into maintenance mode  
> at the end of 2009 and move the bulk of our resources into 
> developing  
> a SIP 3.0 family designed from the ground up to be Draft-Standard  
> achievable.
> 
> 3) RAI Essential Corrections WG: Does this justify its own working  
> group? Remember, it would be not just RFC 3261, but the 
> entire family  
> of related RFCs that would need lock-step maintenance and 
> corrections.  
> Another approach might be to roll the corrections into the Draft  
> Standard version of SIP 2.0.
> 
> 4) RAI Policies WG: We have these ongoing, not-quite-committed work  
> items like session policies, use of SAML, etc.  that we've 
> never been  
> able to get fully wrapped up. I suspect they need their own 
> WG. Or we  
> need to decide that these things just aren't worth doing and stop  
> feeling guilty about not finishing. As it is, I don't have a good  
> feeling about SIP finishing them.
> 
> 5) SIP Operations: I'd like to see SIPPING replaced with a "how to"  
> group possibly under the OPS banner. They could deal with all 
> the "how  
> do we do X with SIP" from other SDOs. When confronted by a need to  
> extend SIP to meet a given need, they could then BOF up a new WG to  
> deal with the required extension.
> 
> We already have working groups looking at conferencing, peering,  
> telephony feature interop, provisioning, peer-to-peer, NAT traversal,
> 
> Other things that might need WGs: SPIT? Identity and Reputation?
> 
> I also suspect we need a couple of directorates that are not WGs to  
> act in an advisory capacity:
> 
> 1) RAI Security Directorate: Focuses on security review and  
> architectural issues in RAI documents.
> 
> 2) RAI Event Package Directorate: Reviews and approves new SIP event  
> packages
> 
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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Subject: Re: [Sip] Toward the Evolution of SIP and Related Working Groups
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> > draft-ietf-sip-saml
> -- I'm not sure this one is ever getting done.
> 
> So I'm not sure there's enough there to justify a WG.
> 
> How about an "Identity in SIP" working group that takes on 
> fixing RFC 4474 for gateways/b2buas and possibly considers 
> identity/role assertion using SAML?

I agree we need that.  I have tried, and so far failed, to
initiate activity towards such an effort.

-d

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From: Dale.Worley@comcast.net
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Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
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   From: "Christer Holmberg" <christer.holmberg@ericsson.com>

   [CHH] Whether the text should be in the document at all depends on if we
   allow 199 to be sent reliably in the first place. Based on the comments
   received so far we should not mandate 199 to be sent reliably, even if
   100rel is required by the UAC. But, the question if whether we want to
   FORBID sending it reliably.

If we ever might allow 199 to be used for HERFP, we should admit the
possibility of sending it reliably in the first draft.  Otherwise,
we'll be locked out of sending it reliably in the future.

Dale
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From sip-bounces@ietf.org  Mon Jun 23 19:01:13 2008
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Subject: Re: [Sip] Toward the Evolution of SIP and Related Working Groups
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On Jun 23, 2008, at 4:48 PM, Dan Wing wrote:

>>> draft-ietf-sip-saml
>> -- I'm not sure this one is ever getting done.
>>
>> So I'm not sure there's enough there to justify a WG.
>>
>> How about an "Identity in SIP" working group that takes on
>> fixing RFC 4474 for gateways/b2buas and possibly considers
>> identity/role assertion using SAML?
>
> I agree we need that.  I have tried, and so far failed, to
> initiate activity towards such an effort.
>

What do you think of a charter of that size, instead of a "All of SIP  
security, forever and ever" as a charter?

--
Dean

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From c.schulz@guldplace.de  Mon Jun 23 19:17:51 2008
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From: "Dharmesh desai" <Dharmesh.Desai@Matrixtelesol.com>
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Subject: [Sip] outbound proxy working with ohter servers
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Dear All,

 

Currently I am developing IP-pbx.

Can any one suggest how out bound proxy works with different servers?

How out bound proxy communicate with registrar server, b2bua?

Any free outbound proxy available so, I get proper idea of its working.

 

Regards,

Dharmesh Desai


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xmlns=3D"http://www.w3.org/TR/REC-html40">

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<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Dear 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'><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'>Currently I am developing =
IP-pbx.<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'>Can any one suggest how out bound proxy works with =
different
servers?<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'>How out bound proxy communicate with registrar =
server,
b2bua?<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'>Any free outbound proxy available so, I get proper =
idea of
its working.<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'>Regards,<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'>Dharmesh Desai<o:p></o:p></span></font></p>

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Just combining both the discussions here.
-----------------------------------------

pkyzivat@cisco.com wrote
>If the call has been forked, and provisional responses with multiple
to-tags have been sent back to the UAC, then it is certain that the UAC
has >*some* resources tied up with each early dialog. The 199 will allow
some of that to be released.

>Are you suggesting that the UAC indicate if it considers those
resources to be sufficient to justify processing an extra message? Or
maybe you   >are just concerned with the likelihood that the UAC doesn't
support 199, so that the message will just be ignored in any case?

> I could go either way on this one.

>	Paul

pkyzivat@cisco.com wrote
> Right - Require is inappropriate. Supported is certainly possible. I
think it would simply mean that the UAC understands the significance of
> 199 and so is prepared to take advantage of it if received. I see no
reason why you would change it on a call by call basis. The resources
aren't > committed at the time of the INVITE - rather at the time the
provisional response is received, so doing on a per-call basis wouldn't
make much > > sense.


[Ashish Saxena] I am concerned about both (justify processing of an
extra message & UAC does not support 199). I brought up this discussion
with the assumption that UAC (mostly) is in better situation to
understand about resources reserved at its end. But Christer pointed out
that even intermediate entities may also be interested in releasing
resources asap. So if an entity knows that it has reserved resources and
would appreciate if they can be relinquished earlier, then it can simply
put "Supported" header to indicate entities downstream that it can
handle 199. This is why I was suggesting it to be call by call basis. 

	But with Paul's reasoning, it will be good to have for all
calls. It is easier to implement also. (otherwise UAC will have to put
extra intelligence to calculate whether it needs 199 in this call).
	
	IMO, Supported header will >

	a) Will stop proxies sending 199 because they would already know
that UAC does not understand it anyways.
	b) Reduce some 199s on network because of (a).



Best Regards,
Ashish Saxena


-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
Sent: Monday, June 23, 2008 9:20 PM
To: Christer Holmberg
Cc: Avasarala Ranjit-A20990; Ashish Saxena (WT01 - Telecom Equipment);
sip@ietf.org
Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00

Right - Require is inappropriate. Supported is certainly possible. I
think it would simply mean that the UAC understands the significance of
199 and so is prepared to take advantage of it if received. I see no
reason why you would change it on a call by call basis. The resources
aren't committed at the time of the INVITE - rather at the time the
provisional response is received, so doing on a per-call basis wouldn't
make much sense.

	Paul

Christer Holmberg wrote:
> Hi, 
> 
>> Can we use Require:199 to indicate to proxy to send 199? 
> 
> No. The Require header is only applicaple to a UAS.
> 
> You can use Proxy-Require for proxies, but the problem is that you
would
> require this from EVERY proxy in the path.
> 
> Regards,
> 
> Christer
> 
> 
> 
> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> ashish.hsaxena@wipro.com
> Sent: Monday, June 23, 2008 10:24 AM
> To: ashish.hsaxena@wipro.com; christer.holmberg@ericsson.com
> Cc: sip@ietf.org
> Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
> 
> 
> Probably everybody is busy. Any thoughts on this one.
> 
> 
> Best Regards,
> Ashish Saxena
> 877-5570
> 
> -----Original Message-----
> From: Ashish Saxena (WT01 - Telecom Equipment)
> Sent: Friday, June 20, 2008 10:56 AM
> To: christer.holmberg@ericsson.com
> Cc: sip@ietf.org
> Subject: RE: [Sip] Draft submission: draft-ietf-sip-199-00
> 
> Hi Christer,
> 	From Section 4 (Client Behaviour) part of this draft, I
> understood that primary objective of this draft is to give indication
to
> UAC to release resources, if any. I am little worried about extra
> messaging that will be seen because of 199 implementation. Can there
be
> a way to tell proxies and UAS downstream that UAC has some resources
> reserved for this call and would be interested in receiving 199, if
> possible?
> 
> IMO, UAC (mostly) is in better understanding of the resources that it
> has reserved for a particular call. So UAC can selectively add
something
> in INVITE to tell downstream proxies/ UAS' about its interest. 
> 
> IMO, we can have a package for 199 and UAC can use "Supported" header
to
> indicate entities downstream that it would be interested in getting
199
> for this *particular* call. Implicitly it would mean that UAC has
> reserved some resources. I understand that we would changing the
meaning
> of "Supported" header on per call basis :-(. Something else may be?
> 
> Comments please.
> 
> Best Regards,
> Ashish Saxena
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use
> sip-implementors@cs.columbia.edu for questions on current sip Use
> sipping@ietf.org for new developments on the application of sip
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
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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" <christer.holmberg@ericsson.com>
To: <ashish.hsaxena@wipro.com>,
	<pkyzivat@cisco.com>
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Hi, 

>Just combining both the discussions here.
>-----------------------------------------
>
>pkyzivat@cisco.com wrote
>>If the call has been forked, and provisional responses with multiple
>>to-tags have been sent back to the UAC, then it is certain that the
UAC has *some* resources tied up with each early 
>>dialog. The 199 will allow some of that to be released.
>>
>>Are you suggesting that the UAC indicate if it considers those
>>resources to be sufficient to justify processing an extra message? Or
>>maybe you are just concerned with the likelihood that the UAC doesn't
>>support 199, so that the message will just be ignored in any case?
>>
>> I could go either way on this one.
>>
>>	Paul
>>
>pkyzivat@cisco.com wrote
>>Right - Require is inappropriate. Supported is certainly possible. I
>>think it would simply mean that the UAC understands the significance
of
>>199 and so is prepared to take advantage of it if received. I see no
>>reason why you would change it on a call by call basis. The resources
aren't committed at the time of the INVITE - 
>>rather at the time the provisional response is received, so doing on a
per-call basis wouldn't make much > > sense.
>
>
>[Ashish Saxena] I am concerned about both (justify processing of an
extra message & UAC does not support 199). I brought 
>up this discussion with the assumption that UAC (mostly) is in better
situation to understand about resources reserved at 
>its end. But Christer pointed out that even intermediate entities may
also be interested in releasing resources asap. So 
>if an entity knows that it has reserved resources and would appreciate
if they can be relinquished earlier, then it can 
>simply put "Supported" header to indicate entities downstream that it
can handle 199. This is why I was suggesting it to 
>be call by call basis. 

[CHH] Sure, if it's ok to say that an intermediate can add an option-tag
to the Supported header.

>But with Paul's reasoning, it will be good to have for all calls. It is
easier to implement also. (otherwise UAC will 
>have to put extra intelligence to calculate whether it needs 199 in
this call).
>	
>	IMO, Supported header will 
>
>	a) Will stop proxies sending 199 because they would already know
that UAC does not understand it anyways.

If an intermediate inserts the option-tag in the Supported header the
UAC would of course still get the 199.

>	b) Reduce some 199s on network because of (a).

[CHH] I don't have any strong feelings whether we define an option-tag
or not.

Regards,

Christer




-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
Sent: Monday, June 23, 2008 9:20 PM
To: Christer Holmberg
Cc: Avasarala Ranjit-A20990; Ashish Saxena (WT01 - Telecom Equipment);
sip@ietf.org
Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00

Right - Require is inappropriate. Supported is certainly possible. I
think it would simply mean that the UAC understands the significance of
199 and so is prepared to take advantage of it if received. I see no
reason why you would change it on a call by call basis. The resources
aren't committed at the time of the INVITE - rather at the time the
provisional response is received, so doing on a per-call basis wouldn't
make much sense.

	Paul

Christer Holmberg wrote:
> Hi,
> 
>> Can we use Require:199 to indicate to proxy to send 199? 
> 
> No. The Require header is only applicaple to a UAS.
> 
> You can use Proxy-Require for proxies, but the problem is that you
would
> require this from EVERY proxy in the path.
> 
> Regards,
> 
> Christer
> 
> 
> 
> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of 
> ashish.hsaxena@wipro.com
> Sent: Monday, June 23, 2008 10:24 AM
> To: ashish.hsaxena@wipro.com; christer.holmberg@ericsson.com
> Cc: sip@ietf.org
> Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
> 
> 
> Probably everybody is busy. Any thoughts on this one.
> 
> 
> Best Regards,
> Ashish Saxena
> 877-5570
> 
> -----Original Message-----
> From: Ashish Saxena (WT01 - Telecom Equipment)
> Sent: Friday, June 20, 2008 10:56 AM
> To: christer.holmberg@ericsson.com
> Cc: sip@ietf.org
> Subject: RE: [Sip] Draft submission: draft-ietf-sip-199-00
> 
> Hi Christer,
> 	From Section 4 (Client Behaviour) part of this draft, I
understood 
> that primary objective of this draft is to give indication
to
> UAC to release resources, if any. I am little worried about extra 
> messaging that will be seen because of 199 implementation. Can there
be
> a way to tell proxies and UAS downstream that UAC has some resources 
> reserved for this call and would be interested in receiving 199, if 
> possible?
> 
> IMO, UAC (mostly) is in better understanding of the resources that it 
> has reserved for a particular call. So UAC can selectively add
something
> in INVITE to tell downstream proxies/ UAS' about its interest. 
> 
> IMO, we can have a package for 199 and UAC can use "Supported" header
to
> indicate entities downstream that it would be interested in getting
199
> for this *particular* call. Implicitly it would mean that UAC has 
> reserved some resources. I understand that we would changing the
meaning
> of "Supported" header on per call basis :-(. Something else may be?
> 
> Comments please.
> 
> Best Regards,
> Ashish Saxena
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use 
> sip-implementors@cs.columbia.edu for questions on current sip Use 
> sipping@ietf.org for new developments on the application of sip 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use 
> sip-implementors@cs.columbia.edu for questions on current sip Use 
> sipping@ietf.org for new developments on the application of sip
> 

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Date: Tue, 24 Jun 2008 08:15:12 +0100
From: "Elwell, John" <john.elwell@siemens.com>
In-reply-to: <E6C2E8958BA59A4FB960963D475F7AC30BE3B850A2@mail.acmepacket.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>,
	"DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>, sip@ietf.org
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Cc: Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Sip] Progress draft-holmberg-sip-keep
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Hadriel,

Concerning your "PBX connections a la SIP trunks" use case, I am not
convinced of this. ETSI TISPAN has specified two ways for an IPPBX to
connect to a service provider. One is the so-called subscription-based
approach, where the IPPBX registers with the SP and communicates via an
edge proxy. In this case, why not use SIP-outbound? The other is the
so-called peering-based approach, which is essentially the same as any
SIP "trunk", e.g., proxy-to-proxy, B2BUA-to-B2BUA, proxy-to-gateway.
SIP-outbound does not apply to these situations, and similarly the
keep-alive mechanism is not specified for this cases. The requirements
in SIP-keep do not cover these situations.

Basically I am not sold on the idea of a separate SIP-keep spec - I
don't think it would be the best use of WG time.

John


> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of Hadriel Kaplan
> Sent: 20 June 2008 18:20
> To: DRAGE, Keith (Keith); sip@ietf.org
> Cc: Christer Holmberg
> Subject: Re: [Sip] Progress draft-holmberg-sip-keep
> 
> 
> My 2 cents: it is a useful draft.  Personally, I would like 
> to have it available for two reasons not cited: PBX 
> connections a la SIP trunks, and proxy-proxy connections.  
> Today I think a lot of people are using OPTIONS requests or 
> proprietary means to perform such keep-alives when 
> registration is not appropriate, but it has led to some 
> interop issues and performance concerns in some cases.
> 
> Since the contentious issue of what form the keep-alives take 
> have already been agreed on for outbound, this seems like a 
> simple draft to get done. (famous last words, I know)
> 
> -hadriel
> 
> 
> > -----Original Message-----
> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of
> > DRAGE, Keith (Keith)
> > Sent: Thursday, June 19, 2008 5:31 AM
> > To: sip@ietf.org
> > Cc: Christer Holmberg
> > Subject: [Sip] Progress draft-holmberg-sip-keep
> >
> > (As SIP WG cochair)
> >
> > We have been asked by the author of
> >
> > http://www.ietf.org/internet-drafts/draft-holmberg-sip-keep-01.txt
> >
> > Whether the SIP WG can progress this document.
> >
> > Because this draft arose as a result of the discussion of 
> outbound, and
> > indeed seems to reuse the requirements from outbound, and these
> > requirements never really got handled in the SIPPING WG, it has been
> > agreed with the SIPPING chairs that we will handle this 
> entirely within
> > SIP.
> >
> > Now in order to ask for charter milestones, and indeed when 
> we finally
> > present this to IESG, we will be asked for the level of 
> support in the
> > WG, which is also predicated on does this fix a real 
> problem, or is it
> > just a corner case with limited application. So:
> >
> > QUESTION 1 TO SIP WG: Are the use cases sufficiently important to
> > proceed with this draft? The document states:
> >
> >    Chapter 3.5 of draft-ietf-sip-outbound-13 [I-D.ietf-sip-outbound]
> >    defines two keep-alive techniques.  Even though the keep-alive
> >    techniques are separated from the Outbound mechanism
> >    [I-D.ietf-sip-outbound], it is currently not possible to indicate
> >    support of the keep-alive techniques without also 
> indicating support
> >    for the Outbound mechanism.
> >
> >    The Outbound mechanism is enabled during the UA 
> registration phase.
> >    However, there are use-cases where the UA does not 
> register itself,
> >    but still needs to be able to make calls and maintain 
> NAT bindings
> >    open during the duration of that call.  A typical example is
> >    emergency calls.  There are also cases where entities do 
> not support
> >    the Outbound mechanism, but still want to be able to 
> indicate support
> >    and use the keep-alive techniques defined in 
> [I-D.ietf-sip-outbound].
> >
> > At first sight this is not the most inspiring declaration 
> of the need
> > for the document. Please respond indicating whether you 
> consider this a
> > useful draft, and propose text that you think would be 
> useful in this
> > section. Conversely, if you think this draft is not useful 
> and the WG
> > has other more important things to work on first, please 
> also respond.
> >
> > QUESTION 2 TO SIP WG: Do we have a robust set of requirements for
> > proceeding with this work? The document currently lists:
> >
> >    REQ 1: It MUST be possible for a UA to indicate support 
> of the keep-
> >    alive techniques defined [I-D.ietf-sip-outbound] if the 
> UA supports
> >    only the keep-alive part of [I-D.ietf-sip-outbound].
> >
> >    REQ 2: It MUST be possible for an edge proxy to indicate 
> support of
> >    the keep-alive techniques defined 
> [I-D.ietf-sip-outbound] if the edge
> >    poxy supports only the keep-alive part of 
> [I-D.ietf-sip-outbound].
> >
> > It would be desirable to agree these at the outset, and not 
> revisit them
> > if we continue with the work. So if you require clarification,
> > modification, or addition to these two requirements, then 
> please also
> > response with your questions and proposals.
> >
> > I suggest we would like responses by 30th June 2008 in 
> order to allow
> > the author to revise the document before the deadlines. 
> Please note that
> > we are looking to make this decision on the list within 
> this deadline
> > based on responses received, not leave it until the Dublin meeting.
> >
> > Regards
> >
> > Keith
> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
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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" <christer.holmberg@ericsson.com>
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Hi,

I agree it may be a good idea to not forbid sending it reliably.

I do think it would be good to have text, saying that it can be sent unreli=
able even if reliable responses are required, though, so that proxies aren'=
t forced to terminate PRACKs etc.

Regards,

Christer
 =


-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Dale.=
Worley@comcast.net
Sent: 21. kes=E4kuuta 2008 1:38
To: sip@ietf.org
Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00

   From: "Christer Holmberg" <christer.holmberg@ericsson.com>

   [CHH] Whether the text should be in the document at all depends on if we
   allow 199 to be sent reliably in the first place. Based on the comments
   received so far we should not mandate 199 to be sent reliably, even if
   100rel is required by the UAC. But, the question if whether we want to
   FORBID sending it reliably.

If we ever might allow 199 to be used for HERFP, we should admit the possib=
ility of sending it reliably in the first draft.  Otherwise, we'll be locke=
d out of sending it reliably in the future.

Dale
_______________________________________________
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This list is for NEW development of the core SIP Protocol Use sip-implement=
ors@cs.columbia.edu for questions on current sip Use sipping@ietf.org for n=
ew developments on the application of sip
_______________________________________________
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens.com>,
	"Hadriel Kaplan" <HKaplan@acmepacket.com>,
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Hi John,

SIP-keep CAN be used proxy-to-proxy, B2BUA-to-B2BUA etc. ANY entity (e.g. a=
 proxy) can insert the keep parameter in its Via header, and if the next en=
tity adds a "yes" value the "keep-alives" can be used between those entitie=
s.

Regards,

Christer
 =


-----Original Message-----
From: Elwell, John [mailto:john.elwell@siemens.com] =

Sent: 24. kes=E4kuuta 2008 10:15
To: Hadriel Kaplan; DRAGE, Keith (Keith); sip@ietf.org
Cc: Christer Holmberg
Subject: RE: [Sip] Progress draft-holmberg-sip-keep

Hadriel,

Concerning your "PBX connections a la SIP trunks" use case, I am not convin=
ced of this. ETSI TISPAN has specified two ways for an IPPBX to connect to =
a service provider. One is the so-called subscription-based approach, where=
 the IPPBX registers with the SP and communicates via an edge proxy. In thi=
s case, why not use SIP-outbound? The other is the so-called peering-based =
approach, which is essentially the same as any SIP "trunk", e.g., proxy-to-=
proxy, B2BUA-to-B2BUA, proxy-to-gateway.
SIP-outbound does not apply to these situations, and similarly the keep-ali=
ve mechanism is not specified for this cases. The requirements in SIP-keep =
do not cover these situations.

Basically I am not sold on the idea of a separate SIP-keep spec - I don't t=
hink it would be the best use of WG time.

John


> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =

> Hadriel Kaplan
> Sent: 20 June 2008 18:20
> To: DRAGE, Keith (Keith); sip@ietf.org
> Cc: Christer Holmberg
> Subject: Re: [Sip] Progress draft-holmberg-sip-keep
> =

> =

> My 2 cents: it is a useful draft.  Personally, I would like to have it =

> available for two reasons not cited: PBX connections a la SIP trunks, =

> and proxy-proxy connections.
> Today I think a lot of people are using OPTIONS requests or =

> proprietary means to perform such keep-alives when registration is not =

> appropriate, but it has led to some interop issues and performance =

> concerns in some cases.
> =

> Since the contentious issue of what form the keep-alives take have =

> already been agreed on for outbound, this seems like a simple draft to =

> get done. (famous last words, I know)
> =

> -hadriel
> =

> =

> > -----Original Message-----
> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
> Behalf Of
> > DRAGE, Keith (Keith)
> > Sent: Thursday, June 19, 2008 5:31 AM
> > To: sip@ietf.org
> > Cc: Christer Holmberg
> > Subject: [Sip] Progress draft-holmberg-sip-keep
> >
> > (As SIP WG cochair)
> >
> > We have been asked by the author of
> >
> > http://www.ietf.org/internet-drafts/draft-holmberg-sip-keep-01.txt
> >
> > Whether the SIP WG can progress this document.
> >
> > Because this draft arose as a result of the discussion of
> outbound, and
> > indeed seems to reuse the requirements from outbound, and these =

> > requirements never really got handled in the SIPPING WG, it has been =

> > agreed with the SIPPING chairs that we will handle this
> entirely within
> > SIP.
> >
> > Now in order to ask for charter milestones, and indeed when
> we finally
> > present this to IESG, we will be asked for the level of
> support in the
> > WG, which is also predicated on does this fix a real
> problem, or is it
> > just a corner case with limited application. So:
> >
> > QUESTION 1 TO SIP WG: Are the use cases sufficiently important to =

> > proceed with this draft? The document states:
> >
> >    Chapter 3.5 of draft-ietf-sip-outbound-13 [I-D.ietf-sip-outbound]
> >    defines two keep-alive techniques.  Even though the keep-alive
> >    techniques are separated from the Outbound mechanism
> >    [I-D.ietf-sip-outbound], it is currently not possible to indicate
> >    support of the keep-alive techniques without also
> indicating support
> >    for the Outbound mechanism.
> >
> >    The Outbound mechanism is enabled during the UA
> registration phase.
> >    However, there are use-cases where the UA does not
> register itself,
> >    but still needs to be able to make calls and maintain
> NAT bindings
> >    open during the duration of that call.  A typical example is
> >    emergency calls.  There are also cases where entities do
> not support
> >    the Outbound mechanism, but still want to be able to
> indicate support
> >    and use the keep-alive techniques defined in
> [I-D.ietf-sip-outbound].
> >
> > At first sight this is not the most inspiring declaration
> of the need
> > for the document. Please respond indicating whether you
> consider this a
> > useful draft, and propose text that you think would be
> useful in this
> > section. Conversely, if you think this draft is not useful
> and the WG
> > has other more important things to work on first, please
> also respond.
> >
> > QUESTION 2 TO SIP WG: Do we have a robust set of requirements for =

> > proceeding with this work? The document currently lists:
> >
> >    REQ 1: It MUST be possible for a UA to indicate support
> of the keep-
> >    alive techniques defined [I-D.ietf-sip-outbound] if the
> UA supports
> >    only the keep-alive part of [I-D.ietf-sip-outbound].
> >
> >    REQ 2: It MUST be possible for an edge proxy to indicate
> support of
> >    the keep-alive techniques defined
> [I-D.ietf-sip-outbound] if the edge
> >    poxy supports only the keep-alive part of
> [I-D.ietf-sip-outbound].
> >
> > It would be desirable to agree these at the outset, and not
> revisit them
> > if we continue with the work. So if you require clarification, =

> > modification, or addition to these two requirements, then
> please also
> > response with your questions and proposals.
> >
> > I suggest we would like responses by 30th June 2008 in
> order to allow
> > the author to revise the document before the deadlines. =

> Please note that
> > we are looking to make this decision on the list within
> this deadline
> > based on responses received, not leave it until the Dublin meeting.
> >
> > Regards
> >
> > Keith
> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use =

> > sip-implementors@cs.columbia.edu for questions on current sip Use =

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

> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

_______________________________________________
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From sip-bounces@ietf.org  Tue Jun 24 00:38:12 2008
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From: Jeroen van Bemmel <jbemmel@zonnet.nl>
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Cc: sip@ietf.org
Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
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Christer,

Should the 199 response contain a Contact header? And if yes, in case a =

proxy sends it, should it contain the address of that proxy (since the =

UAS already sent a final response)?

Should we say that a proxy may only generate and send a 199 when it =

receives a final error response on an INVITE client Transaction which =

was in the PROCEEDING state? (i.e. 1xx response was received before, so =

conceptually sending the 199 response is an action associated with the =

transition from PROCEEDING to COMPLETED)

Regards,
Jeroen

Christer Holmberg wrote:
> Hi,
>
> I agree it may be a good idea to not forbid sending it reliably.
>
> I do think it would be good to have text, saying that it can be sent unre=
liable even if reliable responses are required, though, so that proxies are=
n't forced to terminate PRACKs etc.
>
> Regards,
>
> Christer
>  =

>
> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Dal=
e.Worley@comcast.net
> Sent: 21. kes=E4kuuta 2008 1:38
> To: sip@ietf.org
> Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
>
>    From: "Christer Holmberg" <christer.holmberg@ericsson.com>
>
>    [CHH] Whether the text should be in the document at all depends on if =
we
>    allow 199 to be sent reliably in the first place. Based on the comments
>    received so far we should not mandate 199 to be sent reliably, even if
>    100rel is required by the UAC. But, the question if whether we want to
>    FORBID sending it reliably.
>
> If we ever might allow 199 to be used for HERFP, we should admit the poss=
ibility of sending it reliably in the first draft.  Otherwise, we'll be loc=
ked out of sending it reliably in the future.
>
> Dale
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use sip-impleme=
ntors@cs.columbia.edu for questions on current sip Use sipping@ietf.org for=
 new developments on the application of sip
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>
>   =

_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From sip-bounces@ietf.org  Tue Jun 24 00:56:27 2008
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> In a standards organization, 'success' is defined as
> finishing something. 
Maybe the success criteria should not just to produce documents but to 
produce something that actually sees deployment.


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Hi, =


>Should the 199 response contain a Contact header? And if yes, in case a pr=
oxy sends it, should it contain the address of =

>that proxy (since the UAS already sent a final response)?

IF the 199 is sent reliably be the proxy must contain a Contact header cont=
aining the address of the proxy, yes.

>Should we say that a proxy may only generate and send a 199 when it receiv=
es a final error response on an INVITE client =

>Transaction which was in the PROCEEDING state? (i.e. 1xx response was rece=
ived before, so conceptually sending the 199 =

>response is an action associated with the transition from PROCEEDING to CO=
MPLETED)

I am not sure I understand. The idea IS to send 199 when a final error resp=
onse is received by the forking proxy, if a 18x has previously been receive=
d.

Regards,

Christer




Christer Holmberg wrote:
> Hi,
>
> I agree it may be a good idea to not forbid sending it reliably.
>
> I do think it would be good to have text, saying that it can be sent unre=
liable even if reliable responses are required, though, so that proxies are=
n't forced to terminate PRACKs etc.
>
> Regards,
>
> Christer
>  =

>
> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =

> Dale.Worley@comcast.net
> Sent: 21. kes=E4kuuta 2008 1:38
> To: sip@ietf.org
> Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
>
>    From: "Christer Holmberg" <christer.holmberg@ericsson.com>
>
>    [CHH] Whether the text should be in the document at all depends on if =
we
>    allow 199 to be sent reliably in the first place. Based on the comments
>    received so far we should not mandate 199 to be sent reliably, even if
>    100rel is required by the UAC. But, the question if whether we want to
>    FORBID sending it reliably.
>
> If we ever might allow 199 to be used for HERFP, we should admit the poss=
ibility of sending it reliably in the first draft.  Otherwise, we'll be loc=
ked out of sending it reliably in the future.
>
> Dale
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use =

> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use =

> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

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


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Many of the SIP security mechanisms share the same fate: They are far 
ahead of the actual SIP deployment. This is true for SIP Identity, SIP 
CERT, SIP SAML, End-to-End Security, etc. When we started the SAML work 
we looked at what was going on at that time in the HTTP space. Without 
doubt the entire application layer identity management space found a lot 
of excitement. There is a lot of standardization being done and also a 
lot of deployment taking place. With the SIP space that was obviously a 
bit different and deployments today focus largely on voice (and there 
not even on end-to-end SIP-based communication).

For example: Look at what is being used in XMPP. There is no 
equalivalent of SIP Identity -- folks are currently looking into 
providing certificates for server-to-server communication.

Even though they are, from a deployment point of view, ahead they are 
not even close to where we are with our documents.

Ciao
Hannes

PS: I also believe that the SIP Identity case isn't an easy one either. 
The guys that would make use of SIP Identity for a deployment where the 
two SIP proxies talk to each other there would not be a need for SIP 
Identity since you are essentially replicating what TLS provides you at 
a lower layer already. In cases where there many SIP proxies along the 
path SIP Identity would be useful since it provides protection against 
any one of them being malicious. However, the guys who favour such a 
deployment model are the onces that believe very much in the chain of 
trust (hop-by-hop security). They have no interest in using SIP Identity.



Dan Wing wrote:
>>> draft-ietf-sip-saml
>>>       
>> -- I'm not sure this one is ever getting done.
>>
>> So I'm not sure there's enough there to justify a WG.
>>
>> How about an "Identity in SIP" working group that takes on 
>> fixing RFC 4474 for gateways/b2buas and possibly considers 
>> identity/role assertion using SAML?
>>     
>
> I agree we need that.  I have tried, and so far failed, to
> initiate activity towards such an effort.
>
> -d
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>   

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Date: Tue, 24 Jun 2008 11:05:03 +0100
From: "Elwell, John" <john.elwell@siemens.com>
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	<CA9998CD4A020D418654FCDEF4E707DF06D3A442@esealmw113.eemea.ericsson.se>
To: Christer Holmberg <christer.holmberg@ericsson.com>,
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	"DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>, sip@ietf.org
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Subject: Re: [Sip] Progress draft-holmberg-sip-keep
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Christer,

The majority of the specification talks about a UA as client and an edge pr=
oxy as server. The only hint that it might be applied in other situations i=
s in section 8 "Overlap with connection reuse", but there is no normative l=
anguage there. So the scope clearly seems to be between a UA and an edge pr=
oxy, and I don't see much value in that (i.e., might as well support sip-ou=
tbound).

If you wish to extend the scope so that it can operate between any two SIP =
entities, and in particular between two proxies, then that is a different m=
atter. There may be valid use cases, but I can't recall whether these have =
been discussed. Will the same technical solution apply (e.g., in terms of w=
hich side is the client, which side chooses the timer value, what is the de=
fault timer value, etc.)?

I think Keith's questions related to the present draft (01) and I interpret=
 the scope as UA to edge proxy only. If you wish to extend the scope, this =
would require a revised draft before making any decision on adopting as a w=
ork item.

John

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com] =

> Sent: 24 June 2008 08:20
> To: Elwell, John; Hadriel Kaplan; DRAGE, Keith (Keith); sip@ietf.org
> Subject: RE: [Sip] Progress draft-holmberg-sip-keep
> =

> =

> Hi John,
> =

> SIP-keep CAN be used proxy-to-proxy, B2BUA-to-B2BUA etc. ANY =

> entity (e.g. a proxy) can insert the keep parameter in its =

> Via header, and if the next entity adds a "yes" value the =

> "keep-alives" can be used between those entities.
> =

> Regards,
> =

> Christer
>  =

> =

> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens.com] =

> Sent: 24. kes=E4kuuta 2008 10:15
> To: Hadriel Kaplan; DRAGE, Keith (Keith); sip@ietf.org
> Cc: Christer Holmberg
> Subject: RE: [Sip] Progress draft-holmberg-sip-keep
> =

> Hadriel,
> =

> Concerning your "PBX connections a la SIP trunks" use case, I =

> am not convinced of this. ETSI TISPAN has specified two ways =

> for an IPPBX to connect to a service provider. One is the =

> so-called subscription-based approach, where the IPPBX =

> registers with the SP and communicates via an edge proxy. In =

> this case, why not use SIP-outbound? The other is the =

> so-called peering-based approach, which is essentially the =

> same as any SIP "trunk", e.g., proxy-to-proxy, =

> B2BUA-to-B2BUA, proxy-to-gateway.
> SIP-outbound does not apply to these situations, and =

> similarly the keep-alive mechanism is not specified for this =

> cases. The requirements in SIP-keep do not cover these situations.
> =

> Basically I am not sold on the idea of a separate SIP-keep =

> spec - I don't think it would be the best use of WG time.
> =

> John
> =

> =

> > -----Original Message-----
> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On =

> Behalf Of =

> > Hadriel Kaplan
> > Sent: 20 June 2008 18:20
> > To: DRAGE, Keith (Keith); sip@ietf.org
> > Cc: Christer Holmberg
> > Subject: Re: [Sip] Progress draft-holmberg-sip-keep
> > =

> > =

> > My 2 cents: it is a useful draft.  Personally, I would like =

> to have it =

> > available for two reasons not cited: PBX connections a la =

> SIP trunks, =

> > and proxy-proxy connections.
> > Today I think a lot of people are using OPTIONS requests or =

> > proprietary means to perform such keep-alives when =

> registration is not =

> > appropriate, but it has led to some interop issues and performance =

> > concerns in some cases.
> > =

> > Since the contentious issue of what form the keep-alives take have =

> > already been agreed on for outbound, this seems like a =

> simple draft to =

> > get done. (famous last words, I know)
> > =

> > -hadriel
> > =

> > =

> > > -----Original Message-----
> > > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
> > Behalf Of
> > > DRAGE, Keith (Keith)
> > > Sent: Thursday, June 19, 2008 5:31 AM
> > > To: sip@ietf.org
> > > Cc: Christer Holmberg
> > > Subject: [Sip] Progress draft-holmberg-sip-keep
> > >
> > > (As SIP WG cochair)
> > >
> > > We have been asked by the author of
> > >
> > > http://www.ietf.org/internet-drafts/draft-holmberg-sip-keep-01.txt
> > >
> > > Whether the SIP WG can progress this document.
> > >
> > > Because this draft arose as a result of the discussion of
> > outbound, and
> > > indeed seems to reuse the requirements from outbound, and these =

> > > requirements never really got handled in the SIPPING WG, =

> it has been =

> > > agreed with the SIPPING chairs that we will handle this
> > entirely within
> > > SIP.
> > >
> > > Now in order to ask for charter milestones, and indeed when
> > we finally
> > > present this to IESG, we will be asked for the level of
> > support in the
> > > WG, which is also predicated on does this fix a real
> > problem, or is it
> > > just a corner case with limited application. So:
> > >
> > > QUESTION 1 TO SIP WG: Are the use cases sufficiently important to =

> > > proceed with this draft? The document states:
> > >
> > >    Chapter 3.5 of draft-ietf-sip-outbound-13 =

> [I-D.ietf-sip-outbound]
> > >    defines two keep-alive techniques.  Even though the keep-alive
> > >    techniques are separated from the Outbound mechanism
> > >    [I-D.ietf-sip-outbound], it is currently not possible =

> to indicate
> > >    support of the keep-alive techniques without also
> > indicating support
> > >    for the Outbound mechanism.
> > >
> > >    The Outbound mechanism is enabled during the UA
> > registration phase.
> > >    However, there are use-cases where the UA does not
> > register itself,
> > >    but still needs to be able to make calls and maintain
> > NAT bindings
> > >    open during the duration of that call.  A typical example is
> > >    emergency calls.  There are also cases where entities do
> > not support
> > >    the Outbound mechanism, but still want to be able to
> > indicate support
> > >    and use the keep-alive techniques defined in
> > [I-D.ietf-sip-outbound].
> > >
> > > At first sight this is not the most inspiring declaration
> > of the need
> > > for the document. Please respond indicating whether you
> > consider this a
> > > useful draft, and propose text that you think would be
> > useful in this
> > > section. Conversely, if you think this draft is not useful
> > and the WG
> > > has other more important things to work on first, please
> > also respond.
> > >
> > > QUESTION 2 TO SIP WG: Do we have a robust set of requirements for =

> > > proceeding with this work? The document currently lists:
> > >
> > >    REQ 1: It MUST be possible for a UA to indicate support
> > of the keep-
> > >    alive techniques defined [I-D.ietf-sip-outbound] if the
> > UA supports
> > >    only the keep-alive part of [I-D.ietf-sip-outbound].
> > >
> > >    REQ 2: It MUST be possible for an edge proxy to indicate
> > support of
> > >    the keep-alive techniques defined
> > [I-D.ietf-sip-outbound] if the edge
> > >    poxy supports only the keep-alive part of
> > [I-D.ietf-sip-outbound].
> > >
> > > It would be desirable to agree these at the outset, and not
> > revisit them
> > > if we continue with the work. So if you require clarification, =

> > > modification, or addition to these two requirements, then
> > please also
> > > response with your questions and proposals.
> > >
> > > I suggest we would like responses by 30th June 2008 in
> > order to allow
> > > the author to revise the document before the deadlines. =

> > Please note that
> > > we are looking to make this decision on the list within
> > this deadline
> > > based on responses received, not leave it until the =

> Dublin meeting.
> > >
> > > Regards
> > >
> > > Keith
> > > _______________________________________________
> > > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol Use =

> > > sip-implementors@cs.columbia.edu for questions on current sip Use =

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

> > sip-implementors@cs.columbia.edu for questions on current sip Use =

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

> =

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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens.com>,
	"Hadriel Kaplan" <HKaplan@acmepacket.com>,
	"DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>, <sip@ietf.org>
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Hi, =


>The majority of the specification talks about a UA as client and an edge p=
roxy as server. The only hint that it might be =

>applied in other situations is in section 8 "Overlap with connection reuse=
", but there is no normative language there. So =

>the scope clearly seems to be between a UA and an edge proxy, and I don't =
see much value in that (i.e., might as well =

>support sip-outbound).

Outbound can not always be used, as the draft says.

It is true that initially the primary scope is between the UA and the edge =
proxy. However, I am happy to add the proxy-to-proxy use-case, if people th=
ink that is something we should do.

>If you wish to extend the scope so that it can operate between any two SIP=
 entities, and in particular between two =

>proxies, then that is a different matter. There may be valid use cases, bu=
t I can't recall whether these have been =

>discussed. Will the same technical solution apply (e.g., in terms of which=
 side is the client, which side chooses the =

>timer value, what is the default timer value, etc.)?

At the moment I see no reason why the same technical solution wouldn't appl=
y.

Regards,

Christer



I think Keith's questions related to the present draft (01) and I interpret=
 the scope as UA to edge proxy only. If you =

wish to extend the scope, this would require a revised draft before making =
any decision on adopting as a work item.


John

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: 24 June 2008 08:20
> To: Elwell, John; Hadriel Kaplan; DRAGE, Keith (Keith); sip@ietf.org
> Subject: RE: [Sip] Progress draft-holmberg-sip-keep
> =

> =

> Hi John,
> =

> SIP-keep CAN be used proxy-to-proxy, B2BUA-to-B2BUA etc. ANY entity =

> (e.g. a proxy) can insert the keep parameter in its Via header, and if =

> the next entity adds a "yes" value the "keep-alives" can be used =

> between those entities.
> =

> Regards,
> =

> Christer
>  =

> =

> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens.com]
> Sent: 24. kes=E4kuuta 2008 10:15
> To: Hadriel Kaplan; DRAGE, Keith (Keith); sip@ietf.org
> Cc: Christer Holmberg
> Subject: RE: [Sip] Progress draft-holmberg-sip-keep
> =

> Hadriel,
> =

> Concerning your "PBX connections a la SIP trunks" use case, I am not =

> convinced of this. ETSI TISPAN has specified two ways for an IPPBX to =

> connect to a service provider. One is the so-called subscription-based =

> approach, where the IPPBX registers with the SP and communicates via =

> an edge proxy. In this case, why not use SIP-outbound? The other is =

> the so-called peering-based approach, which is essentially the same as =

> any SIP "trunk", e.g., proxy-to-proxy, B2BUA-to-B2BUA, =

> proxy-to-gateway.
> SIP-outbound does not apply to these situations, and similarly the =

> keep-alive mechanism is not specified for this cases. The requirements =

> in SIP-keep do not cover these situations.
> =

> Basically I am not sold on the idea of a separate SIP-keep spec - I =

> don't think it would be the best use of WG time.
> =

> John
> =

> =

> > -----Original Message-----
> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
> Behalf Of
> > Hadriel Kaplan
> > Sent: 20 June 2008 18:20
> > To: DRAGE, Keith (Keith); sip@ietf.org
> > Cc: Christer Holmberg
> > Subject: Re: [Sip] Progress draft-holmberg-sip-keep
> > =

> > =

> > My 2 cents: it is a useful draft.  Personally, I would like
> to have it
> > available for two reasons not cited: PBX connections a la
> SIP trunks,
> > and proxy-proxy connections.
> > Today I think a lot of people are using OPTIONS requests or =

> > proprietary means to perform such keep-alives when
> registration is not
> > appropriate, but it has led to some interop issues and performance =

> > concerns in some cases.
> > =

> > Since the contentious issue of what form the keep-alives take have =

> > already been agreed on for outbound, this seems like a
> simple draft to
> > get done. (famous last words, I know)
> > =

> > -hadriel
> > =

> > =

> > > -----Original Message-----
> > > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
> > Behalf Of
> > > DRAGE, Keith (Keith)
> > > Sent: Thursday, June 19, 2008 5:31 AM
> > > To: sip@ietf.org
> > > Cc: Christer Holmberg
> > > Subject: [Sip] Progress draft-holmberg-sip-keep
> > >
> > > (As SIP WG cochair)
> > >
> > > We have been asked by the author of
> > >
> > > http://www.ietf.org/internet-drafts/draft-holmberg-sip-keep-01.txt
> > >
> > > Whether the SIP WG can progress this document.
> > >
> > > Because this draft arose as a result of the discussion of
> > outbound, and
> > > indeed seems to reuse the requirements from outbound, and these =

> > > requirements never really got handled in the SIPPING WG,
> it has been
> > > agreed with the SIPPING chairs that we will handle this
> > entirely within
> > > SIP.
> > >
> > > Now in order to ask for charter milestones, and indeed when
> > we finally
> > > present this to IESG, we will be asked for the level of
> > support in the
> > > WG, which is also predicated on does this fix a real
> > problem, or is it
> > > just a corner case with limited application. So:
> > >
> > > QUESTION 1 TO SIP WG: Are the use cases sufficiently important to =

> > > proceed with this draft? The document states:
> > >
> > >    Chapter 3.5 of draft-ietf-sip-outbound-13
> [I-D.ietf-sip-outbound]
> > >    defines two keep-alive techniques.  Even though the keep-alive
> > >    techniques are separated from the Outbound mechanism
> > >    [I-D.ietf-sip-outbound], it is currently not possible
> to indicate
> > >    support of the keep-alive techniques without also
> > indicating support
> > >    for the Outbound mechanism.
> > >
> > >    The Outbound mechanism is enabled during the UA
> > registration phase.
> > >    However, there are use-cases where the UA does not
> > register itself,
> > >    but still needs to be able to make calls and maintain
> > NAT bindings
> > >    open during the duration of that call.  A typical example is
> > >    emergency calls.  There are also cases where entities do
> > not support
> > >    the Outbound mechanism, but still want to be able to
> > indicate support
> > >    and use the keep-alive techniques defined in
> > [I-D.ietf-sip-outbound].
> > >
> > > At first sight this is not the most inspiring declaration
> > of the need
> > > for the document. Please respond indicating whether you
> > consider this a
> > > useful draft, and propose text that you think would be
> > useful in this
> > > section. Conversely, if you think this draft is not useful
> > and the WG
> > > has other more important things to work on first, please
> > also respond.
> > >
> > > QUESTION 2 TO SIP WG: Do we have a robust set of requirements for =

> > > proceeding with this work? The document currently lists:
> > >
> > >    REQ 1: It MUST be possible for a UA to indicate support
> > of the keep-
> > >    alive techniques defined [I-D.ietf-sip-outbound] if the
> > UA supports
> > >    only the keep-alive part of [I-D.ietf-sip-outbound].
> > >
> > >    REQ 2: It MUST be possible for an edge proxy to indicate
> > support of
> > >    the keep-alive techniques defined
> > [I-D.ietf-sip-outbound] if the edge
> > >    poxy supports only the keep-alive part of
> > [I-D.ietf-sip-outbound].
> > >
> > > It would be desirable to agree these at the outset, and not
> > revisit them
> > > if we continue with the work. So if you require clarification, =

> > > modification, or addition to these two requirements, then
> > please also
> > > response with your questions and proposals.
> > >
> > > I suggest we would like responses by 30th June 2008 in
> > order to allow
> > > the author to revise the document before the deadlines. =

> > Please note that
> > > we are looking to make this decision on the list within
> > this deadline
> > > based on responses received, not leave it until the
> Dublin meeting.
> > >
> > > Regards
> > >
> > > Keith
> > > _______________________________________________
> > > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol Use =

> > > sip-implementors@cs.columbia.edu for questions on current sip Use =

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

> > sip-implementors@cs.columbia.edu for questions on current sip Use =

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

> =

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


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Thread-Topic: [Sip] Draft submission: draft-ietf-sip-199-00
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From: "Rockson Li (zhengyli)" <zhengyli@cisco.com>
To: "Christer Holmberg" <christer.holmberg@ericsson.com>,
	"Jeroen van Bemmel" <jbemmel@zonnet.nl>
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Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
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 =

Comment in line
-Rockson
-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Chris=
ter Holmberg
Sent: Tuesday, June 24, 2008 4:05 PM
To: Jeroen van Bemmel
Cc: sip@ietf.org
Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00


Hi, =


>Should the 199 response contain a Contact header? And if yes, in case a =

>proxy sends it, should it contain the address of that proxy (since the UAS=
 already sent a final response)?

IF the 199 is sent reliably be the proxy must contain a Contact header cont=
aining the address of the proxy, yes.

>Should we say that a proxy may only generate and send a 199 when it =

>receives a final error response on an INVITE client Transaction which =

>was in the PROCEEDING state? (i.e. 1xx response was received before, so =

>conceptually sending the 199 response is an action associated with the =

>transition from PROCEEDING to COMPLETED)

I am not sure I understand. The idea IS to send 199 when a final error resp=
onse is received by the forking proxy, if a 18x has previously been receive=
d.
[Rockson] PROCEEDING does not mean early-dialog is established, 100 Trying =
also move  INVITE client Transaction to PROCEEDING state.
                Also, a forking proxy with multiple INVITE client Transacti=
on may receive/forward 180 from one of them, and receive no provisional res=
p and final resp directly
                from the other one, so  INVITE client Transaction's state i=
s not dependable. The decision making must be done in TU.

Regards,

Christer




Christer Holmberg wrote:
> Hi,
>
> I agree it may be a good idea to not forbid sending it reliably.
>
> I do think it would be good to have text, saying that it can be sent unre=
liable even if reliable responses are required, though, so that proxies are=
n't forced to terminate PRACKs etc.
>
> Regards,
>
> Christer
>  =

>
> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =

> Dale.Worley@comcast.net
> Sent: 21. kes=E4kuuta 2008 1:38
> To: sip@ietf.org
> Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
>
>    From: "Christer Holmberg" <christer.holmberg@ericsson.com>
>
>    [CHH] Whether the text should be in the document at all depends on if =
we
>    allow 199 to be sent reliably in the first place. Based on the comments
>    received so far we should not mandate 199 to be sent reliably, even if
>    100rel is required by the UAC. But, the question if whether we want to
>    FORBID sending it reliably.
>
> If we ever might allow 199 to be used for HERFP, we should admit the poss=
ibility of sending it reliably in the first draft.  Otherwise, we'll be loc=
ked out of sending it reliably in the future.
>
> Dale
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use =

> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use =

> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

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


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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Rockson Li (zhengyli)" <zhengyli@cisco.com>,
	"Jeroen van Bemmel" <jbemmel@zonnet.nl>
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Hi, =


>>Should the 199 response contain a Contact header? And if yes, in case a =

>>proxy sends it, should it contain the address of that proxy (since the UA=
S already sent a final response)?
>
>IF the 199 is sent reliably be the proxy must contain a Contact header con=
taining the address of the proxy, yes.
>
>>Should we say that a proxy may only generate and send a 199 when it =

>>receives a final error response on an INVITE client Transaction which =

>>was in the PROCEEDING state? (i.e. 1xx response was received before, so =

>>conceptually sending the 199 response is an action associated with the =

>>transition from PROCEEDING to COMPLETED)
>
>I am not sure I understand. The idea IS to send 199 when a final error res=
ponse is received by the forking proxy, if a =

>18x has previously been received.
>[Rockson] PROCEEDING does not mean early-dialog is established, 100 Trying=
 also move  INVITE client Transaction to =

>PROCEEDING state.

[Christer] 199 is only sent when an early dialog has been established. 100 =
Trying does not establish an early dialog.

Regards,

Christer






Also, a forking proxy with multiple INVITE client Transaction may receive/f=
orward 180 from one of them, and receive no =

provisional resp and final resp directly from the other one, so  INVITE cli=
ent Transaction's state is not dependable. The =

decision making must be done in TU.





Christer Holmberg wrote:
> Hi,
>
> I agree it may be a good idea to not forbid sending it reliably.
>
> I do think it would be good to have text, saying that it can be sent unre=
liable even if reliable responses are required, though, so that proxies are=
n't forced to terminate PRACKs etc.
>
> Regards,
>
> Christer
>  =

>
> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =

> Dale.Worley@comcast.net
> Sent: 21. kes=E4kuuta 2008 1:38
> To: sip@ietf.org
> Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
>
>    From: "Christer Holmberg" <christer.holmberg@ericsson.com>
>
>    [CHH] Whether the text should be in the document at all depends on if =
we
>    allow 199 to be sent reliably in the first place. Based on the comments
>    received so far we should not mandate 199 to be sent reliably, even if
>    100rel is required by the UAC. But, the question if whether we want to
>    FORBID sending it reliably.
>
> If we ever might allow 199 to be used for HERFP, we should admit the poss=
ibility of sending it reliably in the first draft.  Otherwise, we'll be loc=
ked out of sending it reliably in the future.
>
> Dale
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use =

> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use =

> sip-implementors@cs.columbia.edu for questions on current sip Use =

> sipping@ietf.org for new developments on the application of sip
>
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Dean,

Dean Willis wrote:
 >> draft-ietf-sip-ua-privacy
 >
 > -- This one is done,

You might misunderstand the situation of ua-privacy,
but I am still working on that work, and will submit
a new version in a week or so.

An open issue regarding a From header
(what kind of anonymous URI to be set)
was solved in Philadelphia and now I am
studying the treatment of other headers.

Thanks,
Mayumi


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> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of Dan Wing
> Sent: 23 June 2008 22:48
> To: 'Dean Willis'; 'Hadriel Kaplan'
> Cc: sip@ietf.org; 'Paul Kyzivat'
> Subject: Re: [Sip] Toward the Evolution of SIP and Related 
> Working Groups
> 
> > > draft-ietf-sip-saml
> > -- I'm not sure this one is ever getting done.
> > 
> > So I'm not sure there's enough there to justify a WG.
> > 
> > How about an "Identity in SIP" working group that takes on 
> > fixing RFC 4474 for gateways/b2buas and possibly considers 
> > identity/role assertion using SAML?
> 
> I agree we need that.  I have tried, and so far failed, to
> initiate activity towards such an effort.

I think there are enough issues in this area - and yes, they do need
solving - to justify a separate WG, hopefully with a fairly short
lifetime.

John

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Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
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There is a very messy slippery slope here with the reliable 199.

I believe we have agreed that the 199 is to have the to-tag used by the 
UAS. So conceptually it is *from* the UAS. If you want the proxy to send 
the 199 reliably, then it will have to field the response. As others 
have mentioned, this seems to mean that it will have to use its own 
contact address, which will be a change from the contact used previously 
by the UAS for this early dialog.

That gives me heartburn - it just seems wrong. Ignoring that, it puts 
the 199 into the middle of the open question about changing of contact 
addresses and how they take effect.

This could be addressed by always sending the 199 with a To-tag assigned 
by the proxy. But that has its own problems. It actually makes the 
situation worse for UACs that don't support 199.

We sidestep all these problems by not sending the 199 reliably.

Maybe we we should say that the UAC must be prepared to handle both 
reliable and unreliable 199, but for now we don't specify any cases when 
a reliable 199 is sent.

Another thought - there aren't any issues if the *UAS* sends the 199 
reliably. But that isn't going to be the common case.

	Paul

Dale.Worley@comcast.net wrote:
>    From: "Christer Holmberg" <christer.holmberg@ericsson.com>
> 
>    [CHH] Whether the text should be in the document at all depends on if we
>    allow 199 to be sent reliably in the first place. Based on the comments
>    received so far we should not mandate 199 to be sent reliably, even if
>    100rel is required by the UAC. But, the question if whether we want to
>    FORBID sending it reliably.
> 
> If we ever might allow 199 to be used for HERFP, we should admit the
> possibility of sending it reliably in the first draft.  Otherwise,
> we'll be locked out of sending it reliably in the future.
> 
> Dale
> _______________________________________________
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> 
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Hi,

If the proxy assignes its own To-tag, we can't use the To-tag to indciate w=
hich dialog was terminated.

One option, as proposed by Paul, is to say that a UAC shall be able to RECE=
IVE reliable 199 responses.

One ugly solution would be to retransmit the UNRELIABLE 199 a few times - w=
e already use that type of mechanism in ICE, so... ;)

Regards,

Christer


-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Paul =
Kyzivat
Sent: 24. kes=E4kuuta 2008 15:25
To: Dale.Worley@comcast.net
Cc: sip@ietf.org
Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00

There is a very messy slippery slope here with the reliable 199.

I believe we have agreed that the 199 is to have the to-tag used by the UAS=
. So conceptually it is *from* the UAS. If you want the proxy to send the 1=
99 reliably, then it will have to field the response. As others have mentio=
ned, this seems to mean that it will have to use its own contact address, w=
hich will be a change from the contact used previously by the UAS for this =
early dialog.

That gives me heartburn - it just seems wrong. Ignoring that, it puts the 1=
99 into the middle of the open question about changing of contact addresses=
 and how they take effect.

This could be addressed by always sending the 199 with a To-tag assigned by=
 the proxy. But that has its own problems. It actually makes the situation =
worse for UACs that don't support 199.

We sidestep all these problems by not sending the 199 reliably.

Maybe we we should say that the UAC must be prepared to handle both reliabl=
e and unreliable 199, but for now we don't specify any cases when a reliabl=
e 199 is sent.

Another thought - there aren't any issues if the *UAS* sends the 199 reliab=
ly. But that isn't going to be the common case.

	Paul

Dale.Worley@comcast.net wrote:
>    From: "Christer Holmberg" <christer.holmberg@ericsson.com>
> =

>    [CHH] Whether the text should be in the document at all depends on if =
we
>    allow 199 to be sent reliably in the first place. Based on the comments
>    received so far we should not mandate 199 to be sent reliably, even if
>    100rel is required by the UAC. But, the question if whether we want to
>    FORBID sending it reliably.
> =

> If we ever might allow 199 to be used for HERFP, we should admit the =

> possibility of sending it reliably in the first draft.  Otherwise, =

> we'll be locked out of sending it reliably in the future.
> =

> Dale
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use =

> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

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


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From: Paul Kyzivat <pkyzivat@cisco.com>
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Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
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Christer Holmberg wrote:
> Hi,
> =

> If the proxy assignes its own To-tag, we can't use the To-tag to indciate=
 which dialog was terminated.

Right. That was part of my point. In addition, the to-tag it does use =

creates *another* early dialog.

And then, suppose we did allow the proxy to reuse the existing dialog =

for the 199, with a new contact. Then the UAC, upon receipt of the 199, =

will have to send the PRACK within the dialog. So the dialog can't be =

immediately destroyed. When *does* it get destroyed??? When the 200 to =

the PRACK is received?

The extra state management and time required to handle this seems to run =

contrary to the goal of the 199.

> One option, as proposed by Paul, is to say that a UAC shall be able to RE=
CEIVE reliable 199 responses.

Based on my comments above, I am reconsidering the advisability of that.

> One ugly solution would be to retransmit the UNRELIABLE 199 a few times -=
 we already use that type of mechanism in ICE, so... ;)

Yes, that *is* ugly!

	Paul

> Regards,
> =

> Christer
> =

> =

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Pau=
l Kyzivat
> Sent: 24. kes=E4kuuta 2008 15:25
> To: Dale.Worley@comcast.net
> Cc: sip@ietf.org
> Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
> =

> There is a very messy slippery slope here with the reliable 199.
> =

> I believe we have agreed that the 199 is to have the to-tag used by the U=
AS. So conceptually it is *from* the UAS. If you want the proxy to send the=
 199 reliably, then it will have to field the response. As others have ment=
ioned, this seems to mean that it will have to use its own contact address,=
 which will be a change from the contact used previously by the UAS for thi=
s early dialog.
> =

> That gives me heartburn - it just seems wrong. Ignoring that, it puts the=
 199 into the middle of the open question about changing of contact address=
es and how they take effect.
> =

> This could be addressed by always sending the 199 with a To-tag assigned =
by the proxy. But that has its own problems. It actually makes the situatio=
n worse for UACs that don't support 199.
> =

> We sidestep all these problems by not sending the 199 reliably.
> =

> Maybe we we should say that the UAC must be prepared to handle both relia=
ble and unreliable 199, but for now we don't specify any cases when a relia=
ble 199 is sent.
> =

> Another thought - there aren't any issues if the *UAS* sends the 199 reli=
ably. But that isn't going to be the common case.
> =

> 	Paul
> =

> Dale.Worley@comcast.net wrote:
>>    From: "Christer Holmberg" <christer.holmberg@ericsson.com>
>>
>>    [CHH] Whether the text should be in the document at all depends on if=
 we
>>    allow 199 to be sent reliably in the first place. Based on the commen=
ts
>>    received so far we should not mandate 199 to be sent reliably, even if
>>    100rel is required by the UAC. But, the question if whether we want to
>>    FORBID sending it reliably.
>>
>> If we ever might allow 199 to be used for HERFP, we should admit the =

>> possibility of sending it reliably in the first draft.  Otherwise, =

>> we'll be locked out of sending it reliably in the future.
>>
>> Dale
>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use =

>> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

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


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Thread-Topic: [Sip] Draft submission: draft-ietf-sip-199-00
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References: <CA9998CD4A020D418654FCDEF4E707DF06435093@esealmw113.eemea.ericsson.se><200806130255.m5D2t0gC031044@dragon.ariadne.com><CA9998CD4A020D418654FCDEF4E707DF06AF22CF@esealmw113.eemea.ericsson.se>	<200806202237.m5KMbcuk008312@dragon.ariadne.com><CA9998CD4A020D418654FCDEF4E707DF06D3A423@esealmw113.eemea.ericsson.se><4860A45F.9020001@zonnet.nl>
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From: "Rockson Li (zhengyli)" <zhengyli@cisco.com>
To: "Christer Holmberg" <christer.holmberg@ericsson.com>,
	"Jeroen van Bemmel" <jbemmel@zonnet.nl>
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Yes, it's a typo, 100 does not establish early-dialog.
-Rockson =


-----Original Message-----
From: Christer Holmberg [mailto:christer.holmberg@ericsson.com] =

Sent: Tuesday, June 24, 2008 6:32 PM
To: Rockson Li (zhengyli); Jeroen van Bemmel
Cc: sip@ietf.org
Subject: RE: [Sip] Draft submission: draft-ietf-sip-199-00


Hi, =


>>Should the 199 response contain a Contact header? And if yes, in case =

>>a proxy sends it, should it contain the address of that proxy (since the =
UAS already sent a final response)?
>
>IF the 199 is sent reliably be the proxy must contain a Contact header con=
taining the address of the proxy, yes.
>
>>Should we say that a proxy may only generate and send a 199 when it =

>>receives a final error response on an INVITE client Transaction which =

>>was in the PROCEEDING state? (i.e. 1xx response was received before, =

>>so conceptually sending the 199 response is an action associated with =

>>the transition from PROCEEDING to COMPLETED)
>
>I am not sure I understand. The idea IS to send 199 when a final error =

>response is received by the forking proxy, if a 18x has previously been re=
ceived.
>[Rockson] PROCEEDING does not mean early-dialog is established, 100 =

>Trying also move  INVITE client Transaction to PROCEEDING state.

[Christer] 199 is only sent when an early dialog has been established. 100 =
Trying does not establish an early dialog.

Regards,

Christer






Also, a forking proxy with multiple INVITE client Transaction may receive/f=
orward 180 from one of them, and receive no provisional resp and final resp=
 directly from the other one, so  INVITE client Transaction's state is not =
dependable. The decision making must be done in TU.





Christer Holmberg wrote:
> Hi,
>
> I agree it may be a good idea to not forbid sending it reliably.
>
> I do think it would be good to have text, saying that it can be sent unre=
liable even if reliable responses are required, though, so that proxies are=
n't forced to terminate PRACKs etc.
>
> Regards,
>
> Christer
>  =

>
> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =

> Dale.Worley@comcast.net
> Sent: 21. kes=E4kuuta 2008 1:38
> To: sip@ietf.org
> Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
>
>    From: "Christer Holmberg" <christer.holmberg@ericsson.com>
>
>    [CHH] Whether the text should be in the document at all depends on if =
we
>    allow 199 to be sent reliably in the first place. Based on the comments
>    received so far we should not mandate 199 to be sent reliably, even if
>    100rel is required by the UAC. But, the question if whether we want to
>    FORBID sending it reliably.
>
> If we ever might allow 199 to be used for HERFP, we should admit the poss=
ibility of sending it reliably in the first draft.  Otherwise, we'll be loc=
ked out of sending it reliably in the future.
>
> Dale
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use =

> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use =

> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol Use sip-implement=
ors@cs.columbia.edu for questions on current sip Use sipping@ietf.org for n=
ew developments on the application of sip
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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From: Jonathan Rosenberg <jdrosen@cisco.com>
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Subject: Re: [Sip] INFO and what to do about it?
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Mary Barnes wrote:
> I don't believe we could ever forbid INFO. I initially did not think
> we could accomplish anything around INFO, but I believe some of the
> work that's on the table would be useful for working towards
> interoperabilty for the INFO usages. I would be afraid to ask
> honestly for the identification of all the different uses of INFO
> that are out there right now.

I don't think we should be afraid of this at all.

There are (sometimes/often) good reasons why folks resort to these 
solutions. Our job here at IETF is to ensure interoperability for the 
SIP protocol. If we don't listen to our customers - the people who have 
deployed and are actually using it - what purpose does our work serve?


> 
> Doing something is better than nothing at this point IMHO and I'm
> personally really tired of revisiting this issue every couple of
> years. AND, this would help us put a stake in the group on the future
> usages of INFO (whether we ever get rid of the old usages or not), as
> I believe there are other SDOs defining new uses of INFO right now to
> add to the mix of un-interoperability in this area.

As long as SIP usage continues to rise, I suspect we will continue to 
see more INFO usages. Just because we cannot fix what is broken in the 
past, doesn't mean we should let it remain broken for the future.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                   499 Thornall St.
Cisco Fellow                                   Edison, NJ 08837
Cisco, Voice Technology Group
jdrosen@cisco.com
http://www.jdrosen.net                         PHONE: (408) 902-3084
http://www.cisco.com
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Jonathan Rosenberg" <jdrosen@cisco.com>
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Just a couple more points of clarification on my view embedded below
[MB].

Mary

-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@cisco.com] 
Sent: Tuesday, June 24, 2008 8:25 AM
To: Barnes, Mary (RICH2:AR00)
Cc: Christer Holmberg; Robert Sparks; Paul Kyzivat; SIP@ietf.org; DOLLY,
MARTIN C, ATTLABS
Subject: Re: [Sip] INFO and what to do about it?



Mary Barnes wrote:
> I don't believe we could ever forbid INFO. I initially did not think 
> we could accomplish anything around INFO, but I believe some of the 
> work that's on the table would be useful for working towards 
> interoperabilty for the INFO usages. I would be afraid to ask honestly

> for the identification of all the different uses of INFO that are out 
> there right now.

I don't think we should be afraid of this at all.
[MB] My guess (based on what I've seen) is that a lot of vendors would
have around a half dozen (+- 2). Now, some of those might overlap given
there is some level of interop between various vendors. [/MB]

There are (sometimes/often) good reasons why folks resort to these
solutions. Our job here at IETF is to ensure interoperability for the
SIP protocol. If we don't listen to our customers - the people who have
deployed and are actually using it - what purpose does our work serve?

[MB] My experience is that at least half the uses of INFO were in place
when the protocol was quite immature (i.e., well before RFC 3261). And,
we indeed should listen to our customers by providing a flexible
platform for them to do the things they need to do. IMHO, it's been loud
and clear in the past that they want to use INFO and in the past, docs
like "Info considered harmful" weren't helpful towards this end. [/MB]

> 
> Doing something is better than nothing at this point IMHO and I'm 
> personally really tired of revisiting this issue every couple of 
> years. AND, this would help us put a stake in the group on the future 
> usages of INFO (whether we ever get rid of the old usages or not), as 
> I believe there are other SDOs defining new uses of INFO right now to 
> add to the mix of un-interoperability in this area.

As long as SIP usage continues to rise, I suspect we will continue to
see more INFO usages. Just because we cannot fix what is broken in the
past, doesn't mean we should let it remain broken for the future.

[MB] I'm not at all disagreeing on this point. Optimistically, I'd like
to see the current implementations evolve to support this new approach
as it will improve interoperability. If the WG can complete this work in
a timely manner (perhaps the biggest issue given past performance), then
the potential for uptake for a standardized implementation is INFO is
far higher IMHO. If we can get this work chartered, then we're far more
likely to get uptake by the other SDOs far sooner than if we continue to
dillydally. And, it's important to remember that this isn't the only
area where there are interoperability issues. And, actually, per SFSIW-1
the use of INFO was not a common theme - either because it was a given
or it's not as big an issue as some of us are assuming - in a quick scan
I only found one document discussing the use of INFO.  [/MB]

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.                   499 Thornall St.
Cisco Fellow                                   Edison, NJ 08837
Cisco, Voice Technology Group
jdrosen@cisco.com
http://www.jdrosen.net                         PHONE: (408) 902-3084
http://www.cisco.com
_______________________________________________
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Subject: Re: [Sip] INFO and what to do about it?
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Mary,

My latest thought is that we should "grandfather" existing uses of INFO. 
We would provide a registry of them, without blessing them or 
standardizing their specifications. That would at least shine some light 
in the dark corners.

At the same time, we would define the new INFO usage framework (still 
TBD) and ban *new* INFO usages that don't follow it.

I think that would be better than the current situation, and about as 
good as we can expect to achieve.

	Thanks,
	Paul

Mary Barnes wrote:
> Just a couple more points of clarification on my view embedded below
> [MB].
> 
> Mary
> 
> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com] 
> Sent: Tuesday, June 24, 2008 8:25 AM
> To: Barnes, Mary (RICH2:AR00)
> Cc: Christer Holmberg; Robert Sparks; Paul Kyzivat; SIP@ietf.org; DOLLY,
> MARTIN C, ATTLABS
> Subject: Re: [Sip] INFO and what to do about it?
> 
> 
> 
> Mary Barnes wrote:
>> I don't believe we could ever forbid INFO. I initially did not think 
>> we could accomplish anything around INFO, but I believe some of the 
>> work that's on the table would be useful for working towards 
>> interoperabilty for the INFO usages. I would be afraid to ask honestly
> 
>> for the identification of all the different uses of INFO that are out 
>> there right now.
> 
> I don't think we should be afraid of this at all.
> [MB] My guess (based on what I've seen) is that a lot of vendors would
> have around a half dozen (+- 2). Now, some of those might overlap given
> there is some level of interop between various vendors. [/MB]
> 
> There are (sometimes/often) good reasons why folks resort to these
> solutions. Our job here at IETF is to ensure interoperability for the
> SIP protocol. If we don't listen to our customers - the people who have
> deployed and are actually using it - what purpose does our work serve?
> 
> [MB] My experience is that at least half the uses of INFO were in place
> when the protocol was quite immature (i.e., well before RFC 3261). And,
> we indeed should listen to our customers by providing a flexible
> platform for them to do the things they need to do. IMHO, it's been loud
> and clear in the past that they want to use INFO and in the past, docs
> like "Info considered harmful" weren't helpful towards this end. [/MB]
> 
>> Doing something is better than nothing at this point IMHO and I'm 
>> personally really tired of revisiting this issue every couple of 
>> years. AND, this would help us put a stake in the group on the future 
>> usages of INFO (whether we ever get rid of the old usages or not), as 
>> I believe there are other SDOs defining new uses of INFO right now to 
>> add to the mix of un-interoperability in this area.
> 
> As long as SIP usage continues to rise, I suspect we will continue to
> see more INFO usages. Just because we cannot fix what is broken in the
> past, doesn't mean we should let it remain broken for the future.
> 
> [MB] I'm not at all disagreeing on this point. Optimistically, I'd like
> to see the current implementations evolve to support this new approach
> as it will improve interoperability. If the WG can complete this work in
> a timely manner (perhaps the biggest issue given past performance), then
> the potential for uptake for a standardized implementation is INFO is
> far higher IMHO. If we can get this work chartered, then we're far more
> likely to get uptake by the other SDOs far sooner than if we continue to
> dillydally. And, it's important to remember that this isn't the only
> area where there are interoperability issues. And, actually, per SFSIW-1
> the use of INFO was not a common theme - either because it was a given
> or it's not as big an issue as some of us are assuming - in a quick scan
> I only found one document discussing the use of INFO.  [/MB]
> 
> -Jonathan R.
> 
> 
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


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Thread-Topic: [Sip] INFO and what to do about it?
Thread-Index: AcjWApNl36Mi3zFaS9y9KtxcGI9m5gAADdHQ
References: <6BFBFA7D-F646-4647-A084-DF383B6A8371@softarmor.com><F66D7286825402429571678A16C2F5EE03FDC938@zrc2hxm1.corp.nortel.com><3FC39F18-6598-4EF1-A7B8-808E8BD7951D@softarmor.com><BEE5E46D-0056-40FA-99E9-19B6808C0F5A@cisco.com>
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
Cc: SIP@ietf.org, "DOLLY, MARTIN C, ATTLABS" <mdolly@att.com>,
	Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Sip] INFO and what to do about it?
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Paul,

The grandfathering is aa good  suggestion and likely far more pragmatic
than my optimistic approach that they'll evolve. I will go ahead and
ping our folks and see how they feel about this.  If others will do the
same, I think we'll have some very interesting results.

Thanks,
Mary

-----Original Message-----
From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
Sent: Tuesday, June 24, 2008 9:01 AM
To: Barnes, Mary (RICH2:AR00)
Cc: Jonathan Rosenberg; Christer Holmberg; Robert Sparks; SIP@ietf.org;
DOLLY, MARTIN C, ATTLABS
Subject: Re: [Sip] INFO and what to do about it?

Mary,

My latest thought is that we should "grandfather" existing uses of INFO.

We would provide a registry of them, without blessing them or
standardizing their specifications. That would at least shine some light
in the dark corners.

At the same time, we would define the new INFO usage framework (still
TBD) and ban *new* INFO usages that don't follow it.

I think that would be better than the current situation, and about as
good as we can expect to achieve.

	Thanks,
	Paul

Mary Barnes wrote:
> Just a couple more points of clarification on my view embedded below 
> [MB].
> 
> Mary
> 
> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]
> Sent: Tuesday, June 24, 2008 8:25 AM
> To: Barnes, Mary (RICH2:AR00)
> Cc: Christer Holmberg; Robert Sparks; Paul Kyzivat; SIP@ietf.org; 
> DOLLY, MARTIN C, ATTLABS
> Subject: Re: [Sip] INFO and what to do about it?
> 
> 
> 
> Mary Barnes wrote:
>> I don't believe we could ever forbid INFO. I initially did not think 
>> we could accomplish anything around INFO, but I believe some of the 
>> work that's on the table would be useful for working towards 
>> interoperabilty for the INFO usages. I would be afraid to ask 
>> honestly
> 
>> for the identification of all the different uses of INFO that are out

>> there right now.
> 
> I don't think we should be afraid of this at all.
> [MB] My guess (based on what I've seen) is that a lot of vendors would

> have around a half dozen (+- 2). Now, some of those might overlap 
> given there is some level of interop between various vendors. [/MB]
> 
> There are (sometimes/often) good reasons why folks resort to these 
> solutions. Our job here at IETF is to ensure interoperability for the 
> SIP protocol. If we don't listen to our customers - the people who 
> have deployed and are actually using it - what purpose does our work
serve?
> 
> [MB] My experience is that at least half the uses of INFO were in 
> place when the protocol was quite immature (i.e., well before RFC 
> 3261). And, we indeed should listen to our customers by providing a 
> flexible platform for them to do the things they need to do. IMHO, 
> it's been loud and clear in the past that they want to use INFO and in

> the past, docs like "Info considered harmful" weren't helpful towards 
> this end. [/MB]
> 
>> Doing something is better than nothing at this point IMHO and I'm 
>> personally really tired of revisiting this issue every couple of 
>> years. AND, this would help us put a stake in the group on the future

>> usages of INFO (whether we ever get rid of the old usages or not), as

>> I believe there are other SDOs defining new uses of INFO right now to

>> add to the mix of un-interoperability in this area.
> 
> As long as SIP usage continues to rise, I suspect we will continue to 
> see more INFO usages. Just because we cannot fix what is broken in the

> past, doesn't mean we should let it remain broken for the future.
> 
> [MB] I'm not at all disagreeing on this point. Optimistically, I'd 
> like to see the current implementations evolve to support this new 
> approach as it will improve interoperability. If the WG can complete 
> this work in a timely manner (perhaps the biggest issue given past 
> performance), then the potential for uptake for a standardized 
> implementation is INFO is far higher IMHO. If we can get this work 
> chartered, then we're far more likely to get uptake by the other SDOs 
> far sooner than if we continue to dillydally. And, it's important to 
> remember that this isn't the only area where there are 
> interoperability issues. And, actually, per SFSIW-1 the use of INFO 
> was not a common theme - either because it was a given or it's not as 
> big an issue as some of us are assuming - in a quick scan I only found

> one document discussing the use of INFO.  [/MB]
> 
> -Jonathan R.
> 
> 
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


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Dear all,
what is the status of this draft?
Will there be some consideration of this proposal of defining a keep =
parameter?
will this possibly incorporated into draft-ietf-sip-outbound?
=20
Thanks and Best Regards
=20
Roland
=20

Deutsche Telekom AG=20
Zentrum Technik Einf=FChrung=20
Roland Jesske=20
Gateways, Protokolle, Konvergenz, TE32-1
Heinrich-Hertz-Str. 3-7,  64295 Darmstadt, =20
Postfach, 64307 Darmstadt (Postanschrift)=20
+496151 628 2766 (Tel)
+491718618445  (Mobile)=20
E-Mail: r.jesske@telekom.de=20

http://www.telekom.de <http://www.telekom.de/> =20

Registerrechtliche Unternehmensangaben:=20
Deutsche Telekom AG
Aufsichtsrat: Prof. Dr Ulrich Lehner (Vorsitzender)
Vorstand: Ren=E9 Obermann (Vorsitzender),
Dr. Karl-Gerhard Eick (stellvertretender Vorsit-zender),
Hamid Akhavan, Reinhard Clemens, Timotheus H=F6ttges, Thomas =
Sattelberger=20
Handelsregister: Amtsgericht Bonn HRB 6794,=20
Sitz der Gesellschaft: Bonn=20
WEEE-Reg.-Nr.: DE50478376=20
***************************************************************=20
Deutsche Telekom AG=20
Supervisory Board: Prof. Dr Ulrich Lehner (Chairman)=20
Board of Management: Ren=E9 Obermann (Chairman),=20
Dr. Karl-Gerhard Eick (Deputy Chairman),=20
Hamid Akhavan, Reinhard Clemens, Timotheus H=F6ttges,  Thomas =
Sattelberger=20
Commercial register: Amtsgericht Bonn HRB 6794,=20
corporate seat: Bonn=20
WEEEreg.no: DE50478376=20

=20

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<DIV><FONT face=3DArial size=3D2><SPAN class=3D274215314-24062008>Dear=20
all,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D274215314-24062008>what =
is the status=20
of this draft?</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D274215314-24062008>Will =
there be some=20
consideration of this proposal of defining a keep =
parameter?</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D274215314-24062008>will =
this possibly=20
incorporated into draft-ietf-sip-outbound?</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D274215314-24062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D274215314-24062008>Thanks =
and Best=20
Regards</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D274215314-24062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D274215314-24062008>Roland</SPAN></FONT></DIV>
<DIV>&nbsp;</DIV>
<P align=3Dleft><SPAN lang=3Dde><FONT face=3DArial size=3D1>Deutsche =
Telekom=20
AG</FONT></SPAN>&nbsp;<BR><SPAN lang=3Dde><FONT face=3DArial =
size=3D1>Zentrum Technik=20
Einf&#xFC;hrung</FONT></SPAN> <BR><SPAN lang=3Dde><FONT face=3DArial =
size=3D1>Roland=20
Jesske</FONT></SPAN> <BR><SPAN lang=3Dde><FONT face=3DArial =
size=3D1>Gateways,=20
Protokolle, Konvergenz, TE32-1</FONT></SPAN><BR><SPAN lang=3Dde><FONT =
size=3D1><FONT=20
face=3DArial><SPAN lang=3Dde><B>Heinrich-Hertz-Str. =
3-7,</B>&nbsp;</SPAN> 64295=20
Darmstadt,&nbsp;</FONT></FONT></SPAN> <BR><SPAN lang=3Dde><FONT =
face=3DArial=20
size=3D1>Postfach, 64307 Darmstadt =
(Postanschrift)</FONT></SPAN>&nbsp;<BR><FONT=20
face=3DArial size=3D1>+496151 628 2766 (Tel)</FONT><BR><SPAN =
lang=3Dfr><FONT=20
face=3DArial size=3D1>+491718618445 &nbsp;(Mobile)</FONT></SPAN><SPAN=20
lang=3Dde></SPAN> <BR><SPAN lang=3Dfr><FONT face=3DArial =
size=3D1>E-Mail: <A=20
href=3D"mailto:r.jesske@telekom.de">r.jesske@telekom.de</A> =
</FONT></SPAN></P>
<P align=3Dleft><SPAN lang=3Dfr></SPAN><SPAN lang=3Dfr><A=20
href=3D"http://www.telekom.de/"><FONT face=3DArial=20
size=3D2>http://www.telekom.de</FONT></A><FONT face=3DArial =
color=3D#0000ff size=3D2>=20
</FONT></SPAN><SPAN lang=3Dde></P></SPAN><SPAN lang=3Dde></SPAN><SPAN =
lang=3Dde>
<P class=3Deinzug align=3Dleft></SPAN><FONT size=3D1><FONT =
face=3DArial><SPAN=20
lang=3Dde><U>Registerrechtliche Unternehmensangaben:</U></SPAN>=20
<BR></FONT></FONT><SPAN lang=3Dde><FONT size=3D1><FONT =
face=3DArial>Deutsche Telekom=20
AG<BR>Aufsichtsrat:&nbsp;Prof. Dr Ulrich Lehner<SPAN=20
class=3D012202609-06032008><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: black; FONT-FAMILY: Arial; =
mso-fareast-font-family: 'Times New Roman'; mso-ansi-language: DE; =
mso-fareast-language: EN-US; mso-bidi-language: AR-SA">=20
</SPAN></SPAN></FONT></FONT><SPAN class=3D012202609-06032008><FONT =
face=3DArial=20
size=3D1>(Vorsitzender)<BR></FONT></SPAN></SPAN><SPAN lang=3Dde><FONT =
face=3DArial=20
size=3D1>Vorstand: </FONT><FONT face=3DArial size=3D1>Ren&#xE9; Obermann =

(Vorsitzender),<BR></FONT></SPAN><FONT size=3D1><FONT face=3DArial><SPAN =
lang=3Dde>Dr.=20
Karl-Gerhard Eick (stellvertretender Vorsit-zender),<BR>Hamid Akhavan, =
Reinhard=20
Clemens, Timotheus H&#xF6;ttges, Thomas Sattelberger <BR></SPAN><SPAN=20
lang=3Dde>Handelsregister: Amtsgericht Bonn HRB 6794,</SPAN> <BR><SPAN=20
lang=3Dde>Sitz der Gesellschaft: Bonn</SPAN> <BR><SPAN =
lang=3Dde>WEEE-Reg.-Nr.:=20
DE50478376</SPAN> <BR><SPAN=20
lang=3Dde>***************************************************************=
</SPAN>=20
<BR><SPAN lang=3Dde>Deutsche Telekom AG</SPAN> <BR><SPAN =
lang=3Den-gb>Supervisory=20
Board: <SPAN class=3D012202609-06032008>Prof. Dr Ulrich Lehner=20
</SPAN>(Chairman)</SPAN> <BR><SPAN lang=3Den-gb>Board of Management: =
Ren&#xE9; Obermann=20
(Chairman), </SPAN><BR><SPAN lang=3Den-gb>Dr. Karl-Gerhard Eick (Deputy=20
Chairman),</SPAN> <BR><SPAN lang=3Den-gb>Hamid Akhavan, Reinhard =
Clemens,=20
</SPAN><SPAN lang=3Dde>Timotheus H&#xF6;ttges,&nbsp; Thomas =
Sattelberger</SPAN><SPAN=20
lang=3Den-gb></SPAN><SPAN lang=3Den-gb></SPAN> <BR><SPAN =
lang=3Dde>Commercial=20
register: Amtsgericht Bonn HRB 6794,</SPAN> <BR><SPAN =
lang=3Dde>corporate seat:=20
Bonn</SPAN> <BR><SPAN lang=3Dde>WEEEreg.no: =
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Jesske, R" <R.Jesske@telekom.de>,
	<sip@ietf.org>
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Hi Roland,
=20
The latest version of the draft is -01.
=20
http://www.ietf.org/internet-drafts/draft-holmberg-sip-keep-01.txt =
<http://www.ietf.org/internet-drafts/draft-holmberg-sip-keep-01.txt>=20
=20
Keith sent out an e-mail (Subject: Progress draft-holmberg-sip-keep) =
last week (thursday 19th), with some questions for people to answer, in =
order to figure out whether there is interest to progress work with the =
draft. Please take a look.=20
=20
Whatever happens, I am pretty sure it will NOT be incorporated into the =
Outbound spec :)
=20
Regards,
=20
Christer

________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =
Jesske, R
Sent: 24. kes=E4kuuta 2008 18:07
To: sip@ietf.org
Subject: [Sip] Draft: draft-holmberg-sip-keep-00.txt


Dear all,
what is the status of this draft?
Will there be some consideration of this proposal of defining a keep =
parameter?
will this possibly incorporated into draft-ietf-sip-outbound?
=20
Thanks and Best Regards
=20
Roland
=20

Deutsche Telekom AG=20
Zentrum Technik Einf=FChrung=20
Roland Jesske=20
Gateways, Protokolle, Konvergenz, TE32-1
Heinrich-Hertz-Str. 3-7,  64295 Darmstadt, =20
Postfach, 64307 Darmstadt (Postanschrift)=20
+496151 628 2766 (Tel)
+491718618445  (Mobile)=20
E-Mail: r.jesske@telekom.de=20

http://www.telekom.de <http://www.telekom.de/> =20

Registerrechtliche Unternehmensangaben:=20
Deutsche Telekom AG
Aufsichtsrat: Prof. Dr Ulrich Lehner (Vorsitzender)
Vorstand: Ren=E9 Obermann (Vorsitzender),
Dr. Karl-Gerhard Eick (stellvertretender Vorsit-zender),
Hamid Akhavan, Reinhard Clemens, Timotheus H=F6ttges, Thomas =
Sattelberger=20
Handelsregister: Amtsgericht Bonn HRB 6794,=20
Sitz der Gesellschaft: Bonn=20
WEEE-Reg.-Nr.: DE50478376=20
***************************************************************=20
Deutsche Telekom AG=20
Supervisory Board: Prof. Dr Ulrich Lehner (Chairman)=20
Board of Management: Ren=E9 Obermann (Chairman),=20
Dr. Karl-Gerhard Eick (Deputy Chairman),=20
Hamid Akhavan, Reinhard Clemens, Timotheus H=F6ttges,  Thomas =
Sattelberger=20
Commercial register: Amtsgericht Bonn HRB 6794,=20
corporate seat: Bonn=20
WEEEreg.no: DE50478376=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=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2800.1609" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D176291015-24062008><FONT face=3DArial color=3D#0000ff =
size=3D2>Hi=20
Roland,</FONT></SPAN></DIV>
<DIV><SPAN class=3D176291015-24062008><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D176291015-24062008><FONT face=3DArial color=3D#0000ff =
size=3D2>The=20
latest version of the draft is -01.</FONT></SPAN></DIV>
<DIV><SPAN class=3D176291015-24062008><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D176291015-24062008><FONT face=3DArial color=3D#0000ff =
size=3D2><A=20
href=3D"http://www.ietf.org/internet-drafts/draft-holmberg-sip-keep-01.tx=
t"><U><FONT=20
color=3D#0000ff=20
size=3D2>http://www.ietf.org/internet-drafts/draft-holmberg-sip-keep-01.t=
xt</U></FONT></A></DIV></FONT></SPAN>
<DIV><SPAN class=3D176291015-24062008><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D176291015-24062008><FONT face=3DArial color=3D#0000ff =
size=3D2>Keith=20
sent out an e-mail (Subject: Progress draft-holmberg-sip-keep)&nbsp;last =
week=20
(thursday 19th), with some questions for people to answer,&nbsp;in order =
to=20
figure out whether there is interest to progress work with&nbsp;the =
draft.=20
Please take a look. </FONT></SPAN></DIV>
<DIV><SPAN class=3D176291015-24062008><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D176291015-24062008><FONT face=3DArial color=3D#0000ff =

size=3D2>Whatever happens, I am pretty sure it will NOT be incorporated =
into the=20
Outbound spec :)</FONT></SPAN></DIV>
<DIV><SPAN class=3D176291015-24062008><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D176291015-24062008><FONT face=3DArial color=3D#0000ff =

size=3D2>Regards,</FONT></SPAN></DIV>
<DIV><SPAN class=3D176291015-24062008><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D176291015-24062008><FONT face=3DArial color=3D#0000ff =

size=3D2>Christer</FONT></SPAN></DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> sip-bounces@ietf.org=20
[mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Jesske, =
R<BR><B>Sent:</B> 24.=20
kes=E4kuuta 2008 18:07<BR><B>To:</B> sip@ietf.org<BR><B>Subject:</B> =
[Sip] Draft:=20
draft-holmberg-sip-keep-00.txt<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D274215314-24062008>Dear=20
all,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D274215314-24062008>what =
is the status=20
of this draft?</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D274215314-24062008>Will =
there be some=20
consideration of this proposal of defining a keep =
parameter?</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D274215314-24062008>will =
this possibly=20
incorporated into draft-ietf-sip-outbound?</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D274215314-24062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D274215314-24062008>Thanks =
and Best=20
Regards</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D274215314-24062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D274215314-24062008>Roland</SPAN></FONT></DIV>
<DIV>&nbsp;</DIV>
<P align=3Dleft><SPAN lang=3Dde><FONT face=3DArial size=3D1>Deutsche =
Telekom=20
AG</FONT></SPAN>&nbsp;<BR><SPAN lang=3Dde><FONT face=3DArial =
size=3D1>Zentrum Technik=20
Einf=FChrung</FONT></SPAN> <BR><SPAN lang=3Dde><FONT face=3DArial =
size=3D1>Roland=20
Jesske</FONT></SPAN> <BR><SPAN lang=3Dde><FONT face=3DArial =
size=3D1>Gateways,=20
Protokolle, Konvergenz, TE32-1</FONT></SPAN><BR><SPAN lang=3Dde><FONT =
size=3D1><FONT=20
face=3DArial><SPAN lang=3Dde><B>Heinrich-Hertz-Str. =
3-7,</B>&nbsp;</SPAN> 64295=20
Darmstadt,&nbsp;</FONT></FONT></SPAN> <BR><SPAN lang=3Dde><FONT =
face=3DArial=20
size=3D1>Postfach, 64307 Darmstadt =
(Postanschrift)</FONT></SPAN>&nbsp;<BR><FONT=20
face=3DArial size=3D1>+496151 628 2766 (Tel)</FONT><BR><SPAN =
lang=3Dfr><FONT=20
face=3DArial size=3D1>+491718618445 &nbsp;(Mobile)</FONT></SPAN><SPAN=20
lang=3Dde></SPAN> <BR><SPAN lang=3Dfr><FONT face=3DArial =
size=3D1>E-Mail: <A=20
href=3D"mailto:r.jesske@telekom.de">r.jesske@telekom.de</A> =
</FONT></SPAN></P>
<P align=3Dleft><SPAN lang=3Dfr></SPAN><SPAN lang=3Dfr><A=20
href=3D"http://www.telekom.de/"><FONT face=3DArial=20
size=3D2>http://www.telekom.de</FONT></A><FONT face=3DArial =
color=3D#0000ff size=3D2>=20
</FONT></SPAN><SPAN lang=3Dde></P></SPAN><SPAN lang=3Dde></SPAN><SPAN =
lang=3Dde>
<P class=3Deinzug align=3Dleft></SPAN><FONT size=3D1><FONT =
face=3DArial><SPAN=20
lang=3Dde><U>Registerrechtliche Unternehmensangaben:</U></SPAN>=20
<BR></FONT></FONT><SPAN lang=3Dde><FONT size=3D1><FONT =
face=3DArial>Deutsche Telekom=20
AG<BR>Aufsichtsrat:&nbsp;Prof. Dr Ulrich Lehner<SPAN=20
class=3D012202609-06032008><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: black; FONT-FAMILY: Arial; =
mso-fareast-font-family: 'Times New Roman'; mso-ansi-language: DE; =
mso-fareast-language: EN-US; mso-bidi-language: AR-SA">=20
</SPAN></SPAN></FONT></FONT><SPAN class=3D012202609-06032008><FONT =
face=3DArial=20
size=3D1>(Vorsitzender)<BR></FONT></SPAN></SPAN><SPAN lang=3Dde><FONT =
face=3DArial=20
size=3D1>Vorstand: </FONT><FONT face=3DArial size=3D1>Ren=E9 Obermann=20
(Vorsitzender),<BR></FONT></SPAN><FONT size=3D1><FONT face=3DArial><SPAN =
lang=3Dde>Dr.=20
Karl-Gerhard Eick (stellvertretender Vorsit-zender),<BR>Hamid Akhavan, =
Reinhard=20
Clemens, Timotheus H=F6ttges, Thomas Sattelberger <BR></SPAN><SPAN=20
lang=3Dde>Handelsregister: Amtsgericht Bonn HRB 6794,</SPAN> <BR><SPAN=20
lang=3Dde>Sitz der Gesellschaft: Bonn</SPAN> <BR><SPAN =
lang=3Dde>WEEE-Reg.-Nr.:=20
DE50478376</SPAN> <BR><SPAN=20
lang=3Dde>***************************************************************=
</SPAN>=20
<BR><SPAN lang=3Dde>Deutsche Telekom AG</SPAN> <BR><SPAN =
lang=3Den-gb>Supervisory=20
Board: <SPAN class=3D012202609-06032008>Prof. Dr Ulrich Lehner=20
</SPAN>(Chairman)</SPAN> <BR><SPAN lang=3Den-gb>Board of Management: =
Ren=E9 Obermann=20
(Chairman), </SPAN><BR><SPAN lang=3Den-gb>Dr. Karl-Gerhard Eick (Deputy=20
Chairman),</SPAN> <BR><SPAN lang=3Den-gb>Hamid Akhavan, Reinhard =
Clemens,=20
</SPAN><SPAN lang=3Dde>Timotheus H=F6ttges,&nbsp; Thomas =
Sattelberger</SPAN><SPAN=20
lang=3Den-gb></SPAN><SPAN lang=3Den-gb></SPAN> <BR><SPAN =
lang=3Dde>Commercial=20
register: Amtsgericht Bonn HRB 6794,</SPAN> <BR><SPAN =
lang=3Dde>corporate seat:=20
Bonn</SPAN> <BR><SPAN lang=3Dde>WEEEreg.no: =
DE50478376</SPAN></FONT></FONT> </P>
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_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============1805475990==--


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References: <6BFBFA7D-F646-4647-A084-DF383B6A8371@softarmor.com><F66D7286825402429571678A16C2F5EE03FDC938@zrc2hxm1.corp.nortel.com><3FC39F18-6598-4EF1-A7B8-808E8BD7951D@softarmor.com><BEE5E46D-0056-40FA-99E9-19B6808C0F5A@cisco.com>
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Date: Tue, 24 Jun 2008 16:18:27 +0100 (BST)
From: "Michael Procter" <michael@procter.org.uk>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
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Cc: sip@ietf.org, "DOLLY, MARTIN C, ATTLABS" <mdolly@att.com>,
	Mary Barnes <mary.barnes@nortel.com>,
	Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Sip] INFO and what to do about it?
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On Tue, June 24, 2008 3:00 pm, Paul Kyzivat wrote:
> My latest thought is that we should "grandfather" existing uses of INFO.
> We would provide a registry of them, without blessing them or
> standardizing their specifications. That would at least shine some light
> in the dark corners.
>
> At the same time, we would define the new INFO usage framework (still
> TBD) and ban *new* INFO usages that don't follow it.

I agree.  The framework proposed in draft-kaplan-sip-info-events-01 seems
like a step in the right direction.

And I agree with registering existing uses.   Even if existing uses aren't
updated to use the negotiation mechanism suggested (or whatever solution
the WG converges on), I think there is value in strongly encouraging the
adoption of the 'Info-Package' header (or similar).  This shouldn't
noticeably affect interop of legacy devices, but marking the requests like
this would allow more recent devices to act appropriately in the presence
of multiple incompatible non-standard extensions.  Not that such things
exist, of course!

Regards,

Michael

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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
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Hi, =


>>If the proxy assignes its own To-tag, we can't use the To-tag to indciate=
 which dialog was terminated.
>
>Right. That was part of my point. In addition, the to-tag it does use crea=
tes *another* early dialog.
>
>>And then, suppose we did allow the proxy to reuse the existing dialog for=
 the 199, with a new contact. Then the UAC, =

>>upon receipt of the 199, will have to send the PRACK within the dialog. S=
o the dialog can't be immediately destroyed. =

>>When *does* it get destroyed??? When the 200 to the PRACK is received?
>
>The extra state management and time required to handle this seems to run c=
ontrary to the goal of the 199.

Correct.

>>One option, as proposed by Paul, is to say that a UAC shall be able to RE=
CEIVE reliable 199 responses.
>
>Based on my comments above, I am reconsidering the advisability of that.

Well, I guess it wouldn't affect the complexness in the UAC, since it will =
have to be able to handle reliable provisional responses anyway (if it indi=
cates support of it, that is).

And, if the sender of the 199 is a B2BUA, I guess it COULD send it reliably.

But, I have no strong feelings regarding this.

Regards,

Christer






> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =

> Paul Kyzivat
> Sent: 24. kes=E4kuuta 2008 15:25
> To: Dale.Worley@comcast.net
> Cc: sip@ietf.org
> Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
> =

> There is a very messy slippery slope here with the reliable 199.
> =

> I believe we have agreed that the 199 is to have the to-tag used by the U=
AS. So conceptually it is *from* the UAS. If you want the proxy to send the=
 199 reliably, then it will have to field the response. As others have ment=
ioned, this seems to mean that it will have to use its own contact address,=
 which will be a change from the contact used previously by the UAS for thi=
s early dialog.
> =

> That gives me heartburn - it just seems wrong. Ignoring that, it puts the=
 199 into the middle of the open question about changing of contact address=
es and how they take effect.
> =

> This could be addressed by always sending the 199 with a To-tag assigned =
by the proxy. But that has its own problems. It actually makes the situatio=
n worse for UACs that don't support 199.
> =

> We sidestep all these problems by not sending the 199 reliably.
> =

> Maybe we we should say that the UAC must be prepared to handle both relia=
ble and unreliable 199, but for now we don't specify any cases when a relia=
ble 199 is sent.
> =

> Another thought - there aren't any issues if the *UAS* sends the 199 reli=
ably. But that isn't going to be the common case.
> =

> 	Paul
> =

> Dale.Worley@comcast.net wrote:
>>    From: "Christer Holmberg" <christer.holmberg@ericsson.com>
>>
>>    [CHH] Whether the text should be in the document at all depends on if=
 we
>>    allow 199 to be sent reliably in the first place. Based on the commen=
ts
>>    received so far we should not mandate 199 to be sent reliably, even if
>>    100rel is required by the UAC. But, the question if whether we want to
>>    FORBID sending it reliably.
>>
>> If we ever might allow 199 to be used for HERFP, we should admit the =

>> possibility of sending it reliably in the first draft.  Otherwise, =

>> we'll be locked out of sending it reliably in the future.
>>
>> Dale
>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use =

>> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

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Cc: SIP@ietf.org, Paul Kyzivat <pkyzivat@cisco.com>
Subject: Re: [Sip] INFO and what to do about it?
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Hi, all:

It appears that we are almost close to have an agreement on this topic.

The basic analysis needs to be made why people like to use INFO for so many purposes in SIP.

If the WG can abstract those requirements for usages of INFO and can prepare the INFO spec to meet those requirements, there will be no need for "proprietary" usages by vendors.

If the WG fails to meet the above requirements, no matter what is done now or in the future will NOT stop the usage of INFO using proprietary schemes. If the past history provide any lessons, it will be a total failure.

The bottom line is this: The future INFO spec MUST meet the basic requirements for which people are tempted to sue this in the first place.

Best regards,
Radhika

----- Original Message -----
From: Mary Barnes 
Date: Tuesday, June 24, 2008 10:04
Subject: Re: [Sip] INFO and what to do about it?
To: Paul Kyzivat 
Cc: SIP@ietf.org, "DOLLY, MARTIN C, ATTLABS" , Christer Holmberg 

> Paul,
> 
> The grandfathering is aa good suggestion and likely far more 
> pragmaticthan my optimistic approach that they'll evolve. I will 
> go ahead and
> ping our folks and see how they feel about this. If others will 
> do the
> same, I think we'll have some very interesting results.
> 
> Thanks,
> Mary
> 
> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
> Sent: Tuesday, June 24, 2008 9:01 AM
> To: Barnes, Mary (RICH2:AR00)
> Cc: Jonathan Rosenberg; Christer Holmberg; Robert Sparks; 
> SIP@ietf.org;DOLLY, MARTIN C, ATTLABS
> Subject: Re: [Sip] INFO and what to do about it?
> 
> Mary,
> 
> My latest thought is that we should "grandfather" existing uses of 
> INFO.
> We would provide a registry of them, without blessing them or
> standardizing their specifications. That would at least shine some 
> lightin the dark corners.
> 
> At the same time, we would define the new INFO usage framework (still
> TBD) and ban *new* INFO usages that don't follow it.
> 
> I think that would be better than the current situation, and about as
> good as we can expect to achieve.
> 
> Thanks,
> Paul
> 
> Mary Barnes wrote:
> > Just a couple more points of clarification on my view embedded 
> below 
> > [MB].
> > 
> > Mary
> > 
> > -----Original Message-----
> > From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]
> > Sent: Tuesday, June 24, 2008 8:25 AM
> > To: Barnes, Mary (RICH2:AR00)
> > Cc: Christer Holmberg; Robert Sparks; Paul Kyzivat; 
> SIP@ietf.org; 
> > DOLLY, MARTIN C, ATTLABS
> > Subject: Re: [Sip] INFO and what to do about it?
> > 
> > 
> > 
> > Mary Barnes wrote:
> >> I don't believe we could ever forbid INFO. I initially did not 
> think 
> >> we could accomplish anything around INFO, but I believe some of 
> the 
> >> work that's on the table would be useful for working towards 
> >> interoperabilty for the INFO usages. I would be afraid to ask 
> >> honestly
> > 
> >> for the identification of all the different uses of INFO that 
> are out
> 
> >> there right now.
> > 
> > I don't think we should be afraid of this at all.
> > [MB] My guess (based on what I've seen) is that a lot of vendors 
> would
> > have around a half dozen (+- 2). Now, some of those might 
> overlap 
> > given there is some level of interop between various vendors. [/MB]
> > 
> > There are (sometimes/often) good reasons why folks resort to 
> these 
> > solutions. Our job here at IETF is to ensure interoperability 
> for the 
> > SIP protocol. If we don't listen to our customers - the people 
> who 
> > have deployed and are actually using it - what purpose does our work
> serve?
> > 
> > [MB] My experience is that at least half the uses of INFO were 
> in 
> > place when the protocol was quite immature (i.e., well before 
> RFC 
> > 3261). And, we indeed should listen to our customers by 
> providing a 
> > flexible platform for them to do the things they need to do. 
> IMHO, 
> > it's been loud and clear in the past that they want to use INFO 
> and in
> 
> > the past, docs like "Info considered harmful" weren't helpful 
> towards 
> > this end. [/MB]
> > 
> >> Doing something is better than nothing at this point IMHO and 
> I'm 
> >> personally really tired of revisiting this issue every couple 
> of 
> >> years. AND, this would help us put a stake in the group on the 
> future
> >> usages of INFO (whether we ever get rid of the old usages or 
> not), as
> 
> >> I believe there are other SDOs defining new uses of INFO right 
> now to
> 
> >> add to the mix of un-interoperability in this area.
> > 
> > As long as SIP usage continues to rise, I suspect we will 
> continue to 
> > see more INFO usages. Just because we cannot fix what is broken 
> in the
> 
> > past, doesn't mean we should let it remain broken for the future.
> > 
> > [MB] I'm not at all disagreeing on this point. Optimistically, 
> I'd 
> > like to see the current implementations evolve to support this 
> new 
> > approach as it will improve interoperability. If the WG can 
> complete 
> > this work in a timely manner (perhaps the biggest issue given 
> past 
> > performance), then the potential for uptake for a standardized 
> > implementation is INFO is far higher IMHO. If we can get this 
> work 
> > chartered, then we're far more likely to get uptake by the other 
> SDOs 
> > far sooner than if we continue to dillydally. And, it's 
> important to 
> > remember that this isn't the only area where there are 
> > interoperability issues. And, actually, per SFSIW-1 the use of 
> INFO 
> > was not a common theme - either because it was a given or it's 
> not as 
> > big an issue as some of us are assuming - in a quick scan I only 
> found
> > one document discussing the use of INFO. [/MB]
> > 
> > -Jonathan R.
> > 
> > 
> _______________________________________________
> Sip mailing list https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
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Cc: Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Sip] Progress draft-holmberg-sip-keep
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Hey John,
For the case where the IP-PBX registers into the SP, outbound can be used.  But it's the other case I was talking about: where the IP-PBX does NOT register into the SP, but it instead is a trunk.

-hadriel


> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens.com]
> Sent: Tuesday, June 24, 2008 3:15 AM
> To: Hadriel Kaplan; DRAGE, Keith (Keith); sip@ietf.org
> Cc: Christer Holmberg
> Subject: RE: [Sip] Progress draft-holmberg-sip-keep
>
> Hadriel,
>
> Concerning your "PBX connections a la SIP trunks" use case, I am not
> convinced of this. ETSI TISPAN has specified two ways for an IPPBX to
> connect to a service provider. One is the so-called subscription-based
> approach, where the IPPBX registers with the SP and communicates via an
> edge proxy. In this case, why not use SIP-outbound? The other is the
> so-called peering-based approach, which is essentially the same as any
> SIP "trunk", e.g., proxy-to-proxy, B2BUA-to-B2BUA, proxy-to-gateway.
> SIP-outbound does not apply to these situations, and similarly the
> keep-alive mechanism is not specified for this cases. The requirements
> in SIP-keep do not cover these situations.
>
> Basically I am not sold on the idea of a separate SIP-keep spec - I
> don't think it would be the best use of WG time.
>
> John
>
>
> > -----Original Message-----
> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
> > Behalf Of Hadriel Kaplan
> > Sent: 20 June 2008 18:20
> > To: DRAGE, Keith (Keith); sip@ietf.org
> > Cc: Christer Holmberg
> > Subject: Re: [Sip] Progress draft-holmberg-sip-keep
> >
> >
> > My 2 cents: it is a useful draft.  Personally, I would like
> > to have it available for two reasons not cited: PBX
> > connections a la SIP trunks, and proxy-proxy connections.
> > Today I think a lot of people are using OPTIONS requests or
> > proprietary means to perform such keep-alives when
> > registration is not appropriate, but it has led to some
> > interop issues and performance concerns in some cases.
> >
> > Since the contentious issue of what form the keep-alives take
> > have already been agreed on for outbound, this seems like a
> > simple draft to get done. (famous last words, I know)
> >
> > -hadriel
> >
> >
> > > -----Original Message-----
> > > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
> > Behalf Of
> > > DRAGE, Keith (Keith)
> > > Sent: Thursday, June 19, 2008 5:31 AM
> > > To: sip@ietf.org
> > > Cc: Christer Holmberg
> > > Subject: [Sip] Progress draft-holmberg-sip-keep
> > >
> > > (As SIP WG cochair)
> > >
> > > We have been asked by the author of
> > >
> > > http://www.ietf.org/internet-drafts/draft-holmberg-sip-keep-01.txt
> > >
> > > Whether the SIP WG can progress this document.
> > >
> > > Because this draft arose as a result of the discussion of
> > outbound, and
> > > indeed seems to reuse the requirements from outbound, and these
> > > requirements never really got handled in the SIPPING WG, it has been
> > > agreed with the SIPPING chairs that we will handle this
> > entirely within
> > > SIP.
> > >
> > > Now in order to ask for charter milestones, and indeed when
> > we finally
> > > present this to IESG, we will be asked for the level of
> > support in the
> > > WG, which is also predicated on does this fix a real
> > problem, or is it
> > > just a corner case with limited application. So:
> > >
> > > QUESTION 1 TO SIP WG: Are the use cases sufficiently important to
> > > proceed with this draft? The document states:
> > >
> > >    Chapter 3.5 of draft-ietf-sip-outbound-13 [I-D.ietf-sip-outbound]
> > >    defines two keep-alive techniques.  Even though the keep-alive
> > >    techniques are separated from the Outbound mechanism
> > >    [I-D.ietf-sip-outbound], it is currently not possible to indicate
> > >    support of the keep-alive techniques without also
> > indicating support
> > >    for the Outbound mechanism.
> > >
> > >    The Outbound mechanism is enabled during the UA
> > registration phase.
> > >    However, there are use-cases where the UA does not
> > register itself,
> > >    but still needs to be able to make calls and maintain
> > NAT bindings
> > >    open during the duration of that call.  A typical example is
> > >    emergency calls.  There are also cases where entities do
> > not support
> > >    the Outbound mechanism, but still want to be able to
> > indicate support
> > >    and use the keep-alive techniques defined in
> > [I-D.ietf-sip-outbound].
> > >
> > > At first sight this is not the most inspiring declaration
> > of the need
> > > for the document. Please respond indicating whether you
> > consider this a
> > > useful draft, and propose text that you think would be
> > useful in this
> > > section. Conversely, if you think this draft is not useful
> > and the WG
> > > has other more important things to work on first, please
> > also respond.
> > >
> > > QUESTION 2 TO SIP WG: Do we have a robust set of requirements for
> > > proceeding with this work? The document currently lists:
> > >
> > >    REQ 1: It MUST be possible for a UA to indicate support
> > of the keep-
> > >    alive techniques defined [I-D.ietf-sip-outbound] if the
> > UA supports
> > >    only the keep-alive part of [I-D.ietf-sip-outbound].
> > >
> > >    REQ 2: It MUST be possible for an edge proxy to indicate
> > support of
> > >    the keep-alive techniques defined
> > [I-D.ietf-sip-outbound] if the edge
> > >    poxy supports only the keep-alive part of
> > [I-D.ietf-sip-outbound].
> > >
> > > It would be desirable to agree these at the outset, and not
> > revisit them
> > > if we continue with the work. So if you require clarification,
> > > modification, or addition to these two requirements, then
> > please also
> > > response with your questions and proposals.
> > >
> > > I suggest we would like responses by 30th June 2008 in
> > order to allow
> > > the author to revise the document before the deadlines.
> > Please note that
> > > we are looking to make this decision on the list within
> > this deadline
> > > based on responses received, not leave it until the Dublin meeting.
> > >
> > > Regards
> > >
> > > Keith
> > > _______________________________________________
> > > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the application of sip
> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


From sip-bounces@ietf.org  Tue Jun 24 08:48:34 2008
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Date: Tue, 24 Jun 2008 16:48:29 +0100
From: "Elwell, John" <john.elwell@siemens.com>
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	"DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>, sip@ietf.org
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Cc: Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Sip] Progress draft-holmberg-sip-keep
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Hadriel,

That trunking case is no different from other trunking cases, e.g.,
proxy-to-proxy, and I have already suggested to Christer that he might
extend the scope of the draft to cover that. It would be good to have
some discussion on whether keep-alive on "trunking" interfaces would be
beneficial - I have an open mind on that at present.

John 

> -----Original Message-----
> From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com] 
> Sent: 24 June 2008 16:32
> To: Elwell, John; DRAGE, Keith (Keith); sip@ietf.org
> Cc: Christer Holmberg
> Subject: RE: [Sip] Progress draft-holmberg-sip-keep
> 
> Hey John,
> For the case where the IP-PBX registers into the SP, outbound 
> can be used.  But it's the other case I was talking about: 
> where the IP-PBX does NOT register into the SP, but it 
> instead is a trunk.
> 
> -hadriel
> 
> 
> > -----Original Message-----
> > From: Elwell, John [mailto:john.elwell@siemens.com]
> > Sent: Tuesday, June 24, 2008 3:15 AM
> > To: Hadriel Kaplan; DRAGE, Keith (Keith); sip@ietf.org
> > Cc: Christer Holmberg
> > Subject: RE: [Sip] Progress draft-holmberg-sip-keep
> >
> > Hadriel,
> >
> > Concerning your "PBX connections a la SIP trunks" use case, I am not
> > convinced of this. ETSI TISPAN has specified two ways for 
> an IPPBX to
> > connect to a service provider. One is the so-called 
> subscription-based
> > approach, where the IPPBX registers with the SP and 
> communicates via an
> > edge proxy. In this case, why not use SIP-outbound? The other is the
> > so-called peering-based approach, which is essentially the 
> same as any
> > SIP "trunk", e.g., proxy-to-proxy, B2BUA-to-B2BUA, proxy-to-gateway.
> > SIP-outbound does not apply to these situations, and similarly the
> > keep-alive mechanism is not specified for this cases. The 
> requirements
> > in SIP-keep do not cover these situations.
> >
> > Basically I am not sold on the idea of a separate SIP-keep spec - I
> > don't think it would be the best use of WG time.
> >
> > John
> >
> >
> > > -----Original Message-----
> > > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
> > > Behalf Of Hadriel Kaplan
> > > Sent: 20 June 2008 18:20
> > > To: DRAGE, Keith (Keith); sip@ietf.org
> > > Cc: Christer Holmberg
> > > Subject: Re: [Sip] Progress draft-holmberg-sip-keep
> > >
> > >
> > > My 2 cents: it is a useful draft.  Personally, I would like
> > > to have it available for two reasons not cited: PBX
> > > connections a la SIP trunks, and proxy-proxy connections.
> > > Today I think a lot of people are using OPTIONS requests or
> > > proprietary means to perform such keep-alives when
> > > registration is not appropriate, but it has led to some
> > > interop issues and performance concerns in some cases.
> > >
> > > Since the contentious issue of what form the keep-alives take
> > > have already been agreed on for outbound, this seems like a
> > > simple draft to get done. (famous last words, I know)
> > >
> > > -hadriel
> > >
> > >
> > > > -----Original Message-----
> > > > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
> > > Behalf Of
> > > > DRAGE, Keith (Keith)
> > > > Sent: Thursday, June 19, 2008 5:31 AM
> > > > To: sip@ietf.org
> > > > Cc: Christer Holmberg
> > > > Subject: [Sip] Progress draft-holmberg-sip-keep
> > > >
> > > > (As SIP WG cochair)
> > > >
> > > > We have been asked by the author of
> > > >
> > > > 
> http://www.ietf.org/internet-drafts/draft-holmberg-sip-keep-01.txt
> > > >
> > > > Whether the SIP WG can progress this document.
> > > >
> > > > Because this draft arose as a result of the discussion of
> > > outbound, and
> > > > indeed seems to reuse the requirements from outbound, and these
> > > > requirements never really got handled in the SIPPING 
> WG, it has been
> > > > agreed with the SIPPING chairs that we will handle this
> > > entirely within
> > > > SIP.
> > > >
> > > > Now in order to ask for charter milestones, and indeed when
> > > we finally
> > > > present this to IESG, we will be asked for the level of
> > > support in the
> > > > WG, which is also predicated on does this fix a real
> > > problem, or is it
> > > > just a corner case with limited application. So:
> > > >
> > > > QUESTION 1 TO SIP WG: Are the use cases sufficiently 
> important to
> > > > proceed with this draft? The document states:
> > > >
> > > >    Chapter 3.5 of draft-ietf-sip-outbound-13 
> [I-D.ietf-sip-outbound]
> > > >    defines two keep-alive techniques.  Even though the 
> keep-alive
> > > >    techniques are separated from the Outbound mechanism
> > > >    [I-D.ietf-sip-outbound], it is currently not 
> possible to indicate
> > > >    support of the keep-alive techniques without also
> > > indicating support
> > > >    for the Outbound mechanism.
> > > >
> > > >    The Outbound mechanism is enabled during the UA
> > > registration phase.
> > > >    However, there are use-cases where the UA does not
> > > register itself,
> > > >    but still needs to be able to make calls and maintain
> > > NAT bindings
> > > >    open during the duration of that call.  A typical example is
> > > >    emergency calls.  There are also cases where entities do
> > > not support
> > > >    the Outbound mechanism, but still want to be able to
> > > indicate support
> > > >    and use the keep-alive techniques defined in
> > > [I-D.ietf-sip-outbound].
> > > >
> > > > At first sight this is not the most inspiring declaration
> > > of the need
> > > > for the document. Please respond indicating whether you
> > > consider this a
> > > > useful draft, and propose text that you think would be
> > > useful in this
> > > > section. Conversely, if you think this draft is not useful
> > > and the WG
> > > > has other more important things to work on first, please
> > > also respond.
> > > >
> > > > QUESTION 2 TO SIP WG: Do we have a robust set of 
> requirements for
> > > > proceeding with this work? The document currently lists:
> > > >
> > > >    REQ 1: It MUST be possible for a UA to indicate support
> > > of the keep-
> > > >    alive techniques defined [I-D.ietf-sip-outbound] if the
> > > UA supports
> > > >    only the keep-alive part of [I-D.ietf-sip-outbound].
> > > >
> > > >    REQ 2: It MUST be possible for an edge proxy to indicate
> > > support of
> > > >    the keep-alive techniques defined
> > > [I-D.ietf-sip-outbound] if the edge
> > > >    poxy supports only the keep-alive part of
> > > [I-D.ietf-sip-outbound].
> > > >
> > > > It would be desirable to agree these at the outset, and not
> > > revisit them
> > > > if we continue with the work. So if you require clarification,
> > > > modification, or addition to these two requirements, then
> > > please also
> > > > response with your questions and proposals.
> > > >
> > > > I suggest we would like responses by 30th June 2008 in
> > > order to allow
> > > > the author to revise the document before the deadlines.
> > > Please note that
> > > > we are looking to make this decision on the list within
> > > this deadline
> > > > based on responses received, not leave it until the 
> Dublin meeting.
> > > >
> > > > Regards
> > > >
> > > > Keith
> > > > _______________________________________________
> > > > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > > > This list is for NEW development of the core SIP Protocol
> > > > Use sip-implementors@cs.columbia.edu for questions on 
> current sip
> > > > Use sipping@ietf.org for new developments on the 
> application of sip
> > > _______________________________________________
> > > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the 
> application of sip
> > >
> 
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


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On Jun 24, 2008, at 5:56 AM, Mayumi Munakata wrote:

> Dean,
>
> Dean Willis wrote:
> >> draft-ietf-sip-ua-privacy
> >
> > -- This one is done,
>
> You might misunderstand the situation of ua-privacy,
> but I am still working on that work, and will submit
> a new version in a week or so.
>
> An open issue regarding a From header
> (what kind of anonymous URI to be set)
> was solved in Philadelphia and now I am
> studying the treatment of other headers.
>

Ah. I had in fact thought that the From header issue was the main  
issue remaining, and that we had resolved it.

Some sort of incremental discussion of the remaining open issues to  
the mailing list would be very helpful in building us towards a  
consensus. There's no need to wait for a meeting to discuss most  
technical issues.

--
Dean


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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "Elwell, John" <john.elwell@siemens.com>, "DRAGE, Keith (Keith)"
	<drage@alcatel-lucent.com>, "sip@ietf.org" <sip@ietf.org>
Date: Tue, 24 Jun 2008 12:11:21 -0400
Thread-Topic: [Sip] Progress draft-holmberg-sip-keep
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Those cases are the ones I said would be useful in my original email far below.  I.e., when I said "two reasons not cited: PBX connections a la SIP trunks, and proxy-proxy connections."  They are not identical, in the sense that an IP-PBX can be a b2bua/UA/whatever, but also in the sense of their example use-case, for example an IP-PBX could be behind a NAT and thus want this keep-alive for more reasons than a proxy-proxy.

-hadriel

> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens.com]
> Sent: Tuesday, June 24, 2008 11:48 AM
> To: Hadriel Kaplan; DRAGE, Keith (Keith); sip@ietf.org
> Cc: Christer Holmberg
> Subject: RE: [Sip] Progress draft-holmberg-sip-keep
>
> Hadriel,
>
> That trunking case is no different from other trunking cases, e.g.,
> proxy-to-proxy, and I have already suggested to Christer that he might
> extend the scope of the draft to cover that. It would be good to have
> some discussion on whether keep-alive on "trunking" interfaces would be
> beneficial - I have an open mind on that at present.
>
> John
>
> > -----Original Message-----
> > From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]
> > Sent: 24 June 2008 16:32
> > To: Elwell, John; DRAGE, Keith (Keith); sip@ietf.org
> > Cc: Christer Holmberg
> > Subject: RE: [Sip] Progress draft-holmberg-sip-keep
> >
> > Hey John,
> > For the case where the IP-PBX registers into the SP, outbound
> > can be used.  But it's the other case I was talking about:
> > where the IP-PBX does NOT register into the SP, but it
> > instead is a trunk.
> >
> > -hadriel
> >
> >
> > > -----Original Message-----
> > > From: Elwell, John [mailto:john.elwell@siemens.com]
> > > Sent: Tuesday, June 24, 2008 3:15 AM
> > > To: Hadriel Kaplan; DRAGE, Keith (Keith); sip@ietf.org
> > > Cc: Christer Holmberg
> > > Subject: RE: [Sip] Progress draft-holmberg-sip-keep
> > >
> > > Hadriel,
> > >
> > > Concerning your "PBX connections a la SIP trunks" use case, I am not
> > > convinced of this. ETSI TISPAN has specified two ways for
> > an IPPBX to
> > > connect to a service provider. One is the so-called
> > subscription-based
> > > approach, where the IPPBX registers with the SP and
> > communicates via an
> > > edge proxy. In this case, why not use SIP-outbound? The other is the
> > > so-called peering-based approach, which is essentially the
> > same as any
> > > SIP "trunk", e.g., proxy-to-proxy, B2BUA-to-B2BUA, proxy-to-gateway.
> > > SIP-outbound does not apply to these situations, and similarly the
> > > keep-alive mechanism is not specified for this cases. The
> > requirements
> > > in SIP-keep do not cover these situations.
> > >
> > > Basically I am not sold on the idea of a separate SIP-keep spec - I
> > > don't think it would be the best use of WG time.
> > >
> > > John
> > >
> > >
> > > > -----Original Message-----
> > > > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
> > > > Behalf Of Hadriel Kaplan
> > > > Sent: 20 June 2008 18:20
> > > > To: DRAGE, Keith (Keith); sip@ietf.org
> > > > Cc: Christer Holmberg
> > > > Subject: Re: [Sip] Progress draft-holmberg-sip-keep
> > > >
> > > >
> > > > My 2 cents: it is a useful draft.  Personally, I would like
> > > > to have it available for two reasons not cited: PBX
> > > > connections a la SIP trunks, and proxy-proxy connections.
> > > > Today I think a lot of people are using OPTIONS requests or
> > > > proprietary means to perform such keep-alives when
> > > > registration is not appropriate, but it has led to some
> > > > interop issues and performance concerns in some cases.
> > > >
> > > > Since the contentious issue of what form the keep-alives take
> > > > have already been agreed on for outbound, this seems like a
> > > > simple draft to get done. (famous last words, I know)
> > > >
> > > > -hadriel
> > > >
> > > >
> > > > > -----Original Message-----
> > > > > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
> > > > Behalf Of
> > > > > DRAGE, Keith (Keith)
> > > > > Sent: Thursday, June 19, 2008 5:31 AM
> > > > > To: sip@ietf.org
> > > > > Cc: Christer Holmberg
> > > > > Subject: [Sip] Progress draft-holmberg-sip-keep
> > > > >
> > > > > (As SIP WG cochair)
> > > > >
> > > > > We have been asked by the author of
> > > > >
> > > > >
> > http://www.ietf.org/internet-drafts/draft-holmberg-sip-keep-01.txt
> > > > >
> > > > > Whether the SIP WG can progress this document.
> > > > >
> > > > > Because this draft arose as a result of the discussion of
> > > > outbound, and
> > > > > indeed seems to reuse the requirements from outbound, and these
> > > > > requirements never really got handled in the SIPPING
> > WG, it has been
> > > > > agreed with the SIPPING chairs that we will handle this
> > > > entirely within
> > > > > SIP.
> > > > >
> > > > > Now in order to ask for charter milestones, and indeed when
> > > > we finally
> > > > > present this to IESG, we will be asked for the level of
> > > > support in the
> > > > > WG, which is also predicated on does this fix a real
> > > > problem, or is it
> > > > > just a corner case with limited application. So:
> > > > >
> > > > > QUESTION 1 TO SIP WG: Are the use cases sufficiently
> > important to
> > > > > proceed with this draft? The document states:
> > > > >
> > > > >    Chapter 3.5 of draft-ietf-sip-outbound-13
> > [I-D.ietf-sip-outbound]
> > > > >    defines two keep-alive techniques.  Even though the
> > keep-alive
> > > > >    techniques are separated from the Outbound mechanism
> > > > >    [I-D.ietf-sip-outbound], it is currently not
> > possible to indicate
> > > > >    support of the keep-alive techniques without also
> > > > indicating support
> > > > >    for the Outbound mechanism.
> > > > >
> > > > >    The Outbound mechanism is enabled during the UA
> > > > registration phase.
> > > > >    However, there are use-cases where the UA does not
> > > > register itself,
> > > > >    but still needs to be able to make calls and maintain
> > > > NAT bindings
> > > > >    open during the duration of that call.  A typical example is
> > > > >    emergency calls.  There are also cases where entities do
> > > > not support
> > > > >    the Outbound mechanism, but still want to be able to
> > > > indicate support
> > > > >    and use the keep-alive techniques defined in
> > > > [I-D.ietf-sip-outbound].
> > > > >
> > > > > At first sight this is not the most inspiring declaration
> > > > of the need
> > > > > for the document. Please respond indicating whether you
> > > > consider this a
> > > > > useful draft, and propose text that you think would be
> > > > useful in this
> > > > > section. Conversely, if you think this draft is not useful
> > > > and the WG
> > > > > has other more important things to work on first, please
> > > > also respond.
> > > > >
> > > > > QUESTION 2 TO SIP WG: Do we have a robust set of
> > requirements for
> > > > > proceeding with this work? The document currently lists:
> > > > >
> > > > >    REQ 1: It MUST be possible for a UA to indicate support
> > > > of the keep-
> > > > >    alive techniques defined [I-D.ietf-sip-outbound] if the
> > > > UA supports
> > > > >    only the keep-alive part of [I-D.ietf-sip-outbound].
> > > > >
> > > > >    REQ 2: It MUST be possible for an edge proxy to indicate
> > > > support of
> > > > >    the keep-alive techniques defined
> > > > [I-D.ietf-sip-outbound] if the edge
> > > > >    poxy supports only the keep-alive part of
> > > > [I-D.ietf-sip-outbound].
> > > > >
> > > > > It would be desirable to agree these at the outset, and not
> > > > revisit them
> > > > > if we continue with the work. So if you require clarification,
> > > > > modification, or addition to these two requirements, then
> > > > please also
> > > > > response with your questions and proposals.
> > > > >
> > > > > I suggest we would like responses by 30th June 2008 in
> > > > order to allow
> > > > > the author to revise the document before the deadlines.
> > > > Please note that
> > > > > we are looking to make this decision on the list within
> > > > this deadline
> > > > > based on responses received, not leave it until the
> > Dublin meeting.
> > > > >
> > > > > Regards
> > > > >
> > > > > Keith
> > > > > _______________________________________________
> > > > > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > > > > This list is for NEW development of the core SIP Protocol
> > > > > Use sip-implementors@cs.columbia.edu for questions on
> > current sip
> > > > > Use sipping@ietf.org for new developments on the
> > application of sip
> > > > _______________________________________________
> > > > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > > > This list is for NEW development of the core SIP Protocol
> > > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > > Use sipping@ietf.org for new developments on the
> > application of sip
> > > >
> >
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On Jun 24, 2008, at 10:07 AM, Jesske, R wrote:

> Dear all,
> what is the status of this draft?

We are trying to determine whether there is a constituency for the  
draft that justifies further WG effort in advancing it.

> Will there be some consideration of this proposal of defining a keep  
> parameter?

Yes, that's one of the conversations we're having now.

> will this possibly incorporated into draft-ietf-sip-outbound?
>

No. It's more or less been split off from "outbound".

Were I to do this all over again, I would have separated out the  
keepalive function from outbound early on. But we didn't, and outbound  
is pretty much done.

--
Dean

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Cc: SIP@ietf.org, "DOLLY, MARTIN C, ATTLABS" <mdolly@att.com>,
	Mary Barnes <mary.barnes@nortel.com>,
	Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Sip] INFO and what to do about it?
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inline

On Jun 24, 2008, at 9:00 AM, Paul Kyzivat wrote:

> Mary,
>
> My latest thought is that we should "grandfather" existing uses of  
> INFO. We would provide a registry of them, without blessing them or  
> standardizing their specifications. That would at least shine some  
> light in the dark corners.
>
> At the same time, we would define the new INFO usage framework  
> (still TBD) and ban *new* INFO usages that don't follow it.

Well, if we don't think banning INFO as a whole would be listened to,  
why do we think banning INFO that doesn't use this new framework would  
be listened to?

That's where I'm having difficulty with the question. I can see the  
bite of work it would take to make the framework.
I'm still trying to see what would actually use it if we did?
Can somebody list the packages that we've already identified that we  
would put out with this framework?

>
>
> I think that would be better than the current situation, and about  
> as good as we can expect to achieve.

I'm still not convinced it makes it better. One outcome I can see from  
following this path is two years from now finding ourselves with no  
actual packages using the framework and  a set of _new_ INFO based  
applications in the field that _don't_ use the framework.

Will we continue to put those in this proposed registry above? Or do  
we just have another set of practices that we have to treat the  
current non-standardized INFO uses with now?

The bet we would be placing by putting the work into creating this  
framework is that it actually takes some motivation away from using a  
non-standard bare INFO to realize some new application (and that there  
are new applications out there where this would be a really good tool  
to have to realize them). Is this framework really lowering a barrier  
to entry or relieving some other pain so that the people who would  
otherwise make a new non-standard use of INFO come use it? I'm having  
a really hard time seeing that.

If we end up on the wrong side of that bet, we've arguably made things  
_worse_ taking work away from other things, creating mechanics that  
have to be implemented, tested and will ultimately be abused, and not  
solving the actual problem in the first place.

So, I'll continue to look for some evidence this will actually get  
used and relieves some pressure to just go use INFO in a proprietary  
way and skip all this standards pain (hopefully folks here can help me  
with this).

Without it, I think the better path would be to create the registry  
you describe, publish an informational document pointing out the  
pitfalls of the approaches they take, and get on with other work.

I'll also, of course, accept a community decision to just do this work  
and take the risk of losing the bet and help make sure we build the  
best framework possible if that's where we really want to go. I just  
hope somebody can convince me we're not trading off other work we  
could be benefiting from more to find ourselves in effectively the  
same position two years from now.

RjS


>
>
> 	Thanks,
> 	Paul
>
> Mary Barnes wrote:
>> Just a couple more points of clarification on my view embedded below
>> [MB].
>> Mary
>> -----Original Message-----
>> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com] Sent: Tuesday,  
>> June 24, 2008 8:25 AM
>> To: Barnes, Mary (RICH2:AR00)
>> Cc: Christer Holmberg; Robert Sparks; Paul Kyzivat; SIP@ietf.org;  
>> DOLLY,
>> MARTIN C, ATTLABS
>> Subject: Re: [Sip] INFO and what to do about it?
>> Mary Barnes wrote:
>>> I don't believe we could ever forbid INFO. I initially did not  
>>> think we could accomplish anything around INFO, but I believe some  
>>> of the work that's on the table would be useful for working  
>>> towards interoperabilty for the INFO usages. I would be afraid to  
>>> ask honestly
>>> for the identification of all the different uses of INFO that are  
>>> out there right now.
>> I don't think we should be afraid of this at all.
>> [MB] My guess (based on what I've seen) is that a lot of vendors  
>> would
>> have around a half dozen (+- 2). Now, some of those might overlap  
>> given
>> there is some level of interop between various vendors. [/MB]
>> There are (sometimes/often) good reasons why folks resort to these
>> solutions. Our job here at IETF is to ensure interoperability for the
>> SIP protocol. If we don't listen to our customers - the people who  
>> have
>> deployed and are actually using it - what purpose does our work  
>> serve?
>> [MB] My experience is that at least half the uses of INFO were in  
>> place
>> when the protocol was quite immature (i.e., well before RFC 3261).  
>> And,
>> we indeed should listen to our customers by providing a flexible
>> platform for them to do the things they need to do. IMHO, it's been  
>> loud
>> and clear in the past that they want to use INFO and in the past,  
>> docs
>> like "Info considered harmful" weren't helpful towards this end. [/ 
>> MB]
>>> Doing something is better than nothing at this point IMHO and I'm  
>>> personally really tired of revisiting this issue every couple of  
>>> years. AND, this would help us put a stake in the group on the  
>>> future usages of INFO (whether we ever get rid of the old usages  
>>> or not), as I believe there are other SDOs defining new uses of  
>>> INFO right now to add to the mix of un-interoperability in this  
>>> area.
>> As long as SIP usage continues to rise, I suspect we will continue to
>> see more INFO usages. Just because we cannot fix what is broken in  
>> the
>> past, doesn't mean we should let it remain broken for the future.
>> [MB] I'm not at all disagreeing on this point. Optimistically, I'd  
>> like
>> to see the current implementations evolve to support this new  
>> approach
>> as it will improve interoperability. If the WG can complete this  
>> work in
>> a timely manner (perhaps the biggest issue given past performance),  
>> then
>> the potential for uptake for a standardized implementation is INFO is
>> far higher IMHO. If we can get this work chartered, then we're far  
>> more
>> likely to get uptake by the other SDOs far sooner than if we  
>> continue to
>> dillydally. And, it's important to remember that this isn't the only
>> area where there are interoperability issues. And, actually, per  
>> SFSIW-1
>> the use of INFO was not a common theme - either because it was a  
>> given
>> or it's not as big an issue as some of us are assuming - in a quick  
>> scan
>> I only found one document discussing the use of INFO.  [/MB]
>> -Jonathan R.

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Date: Tue, 24 Jun 2008 10:11:11 -0700
From: Henry Sinnreich <hsinnrei@adobe.com>
To: Jonathan Rosenberg <jdrosen@cisco.com>,
	Mary Barnes <mary.barnes@nortel.com>
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Cc: SIP@ietf.org, "DOLLY, MARTIN C, ATTLABS" <mdolly@att.com>,
	Paul Kyzivat <pkyzivat@cisco.com>,
	Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Sip] INFO and what to do about it?
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>If we don't listen to our customers - the people who have
> deployed and are actually using it - what purpose does our work serve?

This is what the managers of the typewriter business must have said when the
word processors appeared: "Our customers ask for a white-out ribbon and for
a price under $500."

A PC at that time cost $5,000 and the additional laser printer was another
$5,000.

Some companies opted for the PC business nevertheless and did NOT listen to
their customers.

Does anyone care for simple SIP that can be understood by developers with
limited time on their hands?

Henry


On 6/24/08 6:25 AM, "Jonathan Rosenberg" <jdrosen@cisco.com> wrote:

> 
> 
> Mary Barnes wrote:
>> I don't believe we could ever forbid INFO. I initially did not think
>> we could accomplish anything around INFO, but I believe some of the
>> work that's on the table would be useful for working towards
>> interoperabilty for the INFO usages. I would be afraid to ask
>> honestly for the identification of all the different uses of INFO
>> that are out there right now.
> 
> I don't think we should be afraid of this at all.
> 
> There are (sometimes/often) good reasons why folks resort to these
> solutions. Our job here at IETF is to ensure interoperability for the
> SIP protocol. If we don't listen to our customers - the people who have
> deployed and are actually using it - what purpose does our work serve?
> 
> 
>> 
>> Doing something is better than nothing at this point IMHO and I'm
>> personally really tired of revisiting this issue every couple of
>> years. AND, this would help us put a stake in the group on the future
>> usages of INFO (whether we ever get rid of the old usages or not), as
>> I believe there are other SDOs defining new uses of INFO right now to
>> add to the mix of un-interoperability in this area.
> 
> As long as SIP usage continues to rise, I suspect we will continue to
> see more INFO usages. Just because we cannot fix what is broken in the
> past, doesn't mean we should let it remain broken for the future.
> 
> -Jonathan R.
> 

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Subject: [Sip] A proposal for breaking the DTLS-SRTP vs RFC4474 gateway
	deadlock
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We've gotten stuck on a fine point in DTLS-SRTP.

The current draft-ietf-sip-dtls-srtp-framework-01 uses an RFC 4474  
Identity header to preserve the integrity of the media key's  
fingerprint, thereby detecting a certain class of MITM attack.

However, RFC 4474 Identity headers are of questionable validity when  
used with protocol gateways or B2BUAs.  More or less, they're capable  
of asserting the identity of the gateway, not the identity of the  
calling party. But the recipient has no real way to figure out which  
is which.

We've debated at some length, and with no good result, about whether  
we should try and fix RFC 4474. We've had some suggestions that may  
work for B2BUAs, and some other suggestions that may work for  
gateways, but we certainly don't have a consensus.

That leaves our chartered deliverable of DTLS-SRTP hanging, and the  
milestone has gone past months ago.

Here's a proposal:

We add a caveat about the limitation of RFC 4474 to draft-ietf-sip- 
dtls-srtp-framework and go ahead and advance that specification. If  
somebody later decides to fix RFC 4474, they can do so, and if  
necessary update DTLS-SRTP if needed.


Does that work for everybody?

If we agree to it, I suggest that we move the date for WGLC of draft- 
ietf-sip-dtls-srtp-framework to July 2008, and move the milestone for  
delivery of that doc to the IESG into September.

--
Dean
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From: "Jean-Francois Mule" <jf.mule@cablelabs.com>
To: "Robert Sparks" <rjsparks@nostrum.com>, "Paul Kyzivat" <pkyzivat@cisco.com>
X-Approved: ondar
Cc: SIP@ietf.org, "DOLLY, MARTIN C, ATTLABS" <mdolly@att.com>,
	Mary Barnes <mary.barnes@nortel.com>,
	Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Sip] INFO and what to do about it?
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I've been reading this thread and so far, it seems that folks are
recommending one or more of the following approaches:
1) do nothing
2) document what's out there in some ways
   (as informational or a new class of WCP RFCs,
worst-current-practices)
3) define new framework for info 
   (as part of the standard track)
   Eric's info draft-02 intended to obsolete 2976 and still showing
intended status as standard track.

The intended audience is both implementers and users (end-users,
enterprise, operators).  
--- implementers
I believe implementers are faced with a basic problem when deciding what
to implement: what other vendor products is my implementation supposed
to interoperate with? Is that other implementation in my customer's
hands.  If so, what is that other vendor implementing for this
pseudo-info functionality.  Correct me if I'm wrong but I think this
often leads to the INFO path for all the uses Eric documented.  Even if
an implementer argues that KPML should be used to pass digits on the
signaling path, if the customer has a box that has INFO for that, they
will likely do info and argue later because there is often little room
for a user/operator to ask for an upgrade of the already deployed box
for a new entrant.
	=> implementers' vote count and what would they choose?
	   Likely:
              1 (because they don't care, those hacks get driven by
customers to interwork with existing gears or they build products in
between UAs...)
	        2 (because they'd like to avoid defining yet a new hack
and would rather code an INFO mechanism to address the maximum number of
implementations out there)
	  Other Opinions?

--- users
While some architects push for pure standard solutions, the reality is
often darker.  Users have often a service or application goal and if it
takes an INFO to get the job done given a box in the field, so be it.
They will try to ask for better solutions in upcoming releases (some of
which never address the INFO hacks which is the problem).
	=> users' opinion count and what would they choose?
	   Likely:
              2 (because they'd like to point to a document and say: I
use this as my UA on my PC/network, and it supports INFO as defined by
that doc section X.Y).
	  Other Opinions?

2 seems at a first sight to be the best worst thing to do.  But doing 2
is a slippery slope for the IETF though: is INFO worth it?  What about
the call diversion headers we can't seem to get rid of in favor of the
their alternative standard-track mechanisms?
It would be good to see a more detailed draft documenting the uses of
INFO (with actual snippets) and some rational numbers indicating how
many implementers (& releases of their code) support those, and how many
users have those.  That could be beneficial to just discuss what to do,
if anything.


Jean-Francois.




> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf
> Of Robert Sparks
> Sent: Tuesday, June 24, 2008 10:52 AM
> To: Paul Kyzivat
> Cc: SIP@ietf.org; DOLLY, MARTIN C, ATTLABS; Mary Barnes; Christer
> Holmberg
> Subject: Re: [Sip] INFO and what to do about it?
> 
> inline
> 
> On Jun 24, 2008, at 9:00 AM, Paul Kyzivat wrote:
> 
> > Mary,
> >
> > My latest thought is that we should "grandfather" existing uses
> of
> > INFO. We would provide a registry of them, without blessing them
> or
> > standardizing their specifications. That would at least shine
> some
> > light in the dark corners.
> >
> > At the same time, we would define the new INFO usage framework
> > (still TBD) and ban *new* INFO usages that don't follow it.
> 
> Well, if we don't think banning INFO as a whole would be listened
> to,
> why do we think banning INFO that doesn't use this new framework
> would
> be listened to?
> 
> That's where I'm having difficulty with the question. I can see the
> bite of work it would take to make the framework.
> I'm still trying to see what would actually use it if we did?
> Can somebody list the packages that we've already identified that
> we
> would put out with this framework?
> 
> >
> >
> > I think that would be better than the current situation, and
> about
> > as good as we can expect to achieve.
> 
> I'm still not convinced it makes it better. One outcome I can see
> from
> following this path is two years from now finding ourselves with no
> actual packages using the framework and  a set of _new_ INFO based
> applications in the field that _don't_ use the framework.
> 
> Will we continue to put those in this proposed registry above? Or
> do
> we just have another set of practices that we have to treat the
> current non-standardized INFO uses with now?
> 
> The bet we would be placing by putting the work into creating this
> framework is that it actually takes some motivation away from using
> a
> non-standard bare INFO to realize some new application (and that
> there
> are new applications out there where this would be a really good
> tool
> to have to realize them). Is this framework really lowering a
> barrier
> to entry or relieving some other pain so that the people who would
> otherwise make a new non-standard use of INFO come use it? I'm
> having
> a really hard time seeing that.
> 
> If we end up on the wrong side of that bet, we've arguably made
> things
> _worse_ taking work away from other things, creating mechanics that
> have to be implemented, tested and will ultimately be abused, and
> not
> solving the actual problem in the first place.
> 
> So, I'll continue to look for some evidence this will actually get
> used and relieves some pressure to just go use INFO in a
> proprietary
> way and skip all this standards pain (hopefully folks here can help
> me
> with this).
> 
> Without it, I think the better path would be to create the registry
> you describe, publish an informational document pointing out the
> pitfalls of the approaches they take, and get on with other work.
> 
> I'll also, of course, accept a community decision to just do this
> work
> and take the risk of losing the bet and help make sure we build the
> best framework possible if that's where we really want to go. I
> just
> hope somebody can convince me we're not trading off other work we
> could be benefiting from more to find ourselves in effectively the
> same position two years from now.
> 
> RjS
> 
> 
> >
> >
> > 	Thanks,
> > 	Paul
> >
> > Mary Barnes wrote:
> >> Just a couple more points of clarification on my view embedded
> below
> >> [MB].
> >> Mary
> >> -----Original Message-----
> >> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com] Sent:
> Tuesday,
> >> June 24, 2008 8:25 AM
> >> To: Barnes, Mary (RICH2:AR00)
> >> Cc: Christer Holmberg; Robert Sparks; Paul Kyzivat;
> SIP@ietf.org;
> >> DOLLY,
> >> MARTIN C, ATTLABS
> >> Subject: Re: [Sip] INFO and what to do about it?
> >> Mary Barnes wrote:
> >>> I don't believe we could ever forbid INFO. I initially did not
> >>> think we could accomplish anything around INFO, but I believe
> some
> >>> of the work that's on the table would be useful for working
> >>> towards interoperabilty for the INFO usages. I would be afraid
> to
> >>> ask honestly
> >>> for the identification of all the different uses of INFO that
> are
> >>> out there right now.
> >> I don't think we should be afraid of this at all.
> >> [MB] My guess (based on what I've seen) is that a lot of vendors
> >> would
> >> have around a half dozen (+- 2). Now, some of those might
> overlap
> >> given
> >> there is some level of interop between various vendors. [/MB]
> >> There are (sometimes/often) good reasons why folks resort to
> these
> >> solutions. Our job here at IETF is to ensure interoperability
> for the
> >> SIP protocol. If we don't listen to our customers - the people
> who
> >> have
> >> deployed and are actually using it - what purpose does our work
> >> serve?
> >> [MB] My experience is that at least half the uses of INFO were
> in
> >> place
> >> when the protocol was quite immature (i.e., well before RFC
> 3261).
> >> And,
> >> we indeed should listen to our customers by providing a flexible
> >> platform for them to do the things they need to do. IMHO, it's
> been
> >> loud
> >> and clear in the past that they want to use INFO and in the
> past,
> >> docs
> >> like "Info considered harmful" weren't helpful towards this end.
> [/
> >> MB]
> >>> Doing something is better than nothing at this point IMHO and
> I'm
> >>> personally really tired of revisiting this issue every couple
> of
> >>> years. AND, this would help us put a stake in the group on the
> >>> future usages of INFO (whether we ever get rid of the old
> usages
> >>> or not), as I believe there are other SDOs defining new uses of
> >>> INFO right now to add to the mix of un-interoperability in this
> >>> area.
> >> As long as SIP usage continues to rise, I suspect we will
> continue to
> >> see more INFO usages. Just because we cannot fix what is broken
> in
> >> the
> >> past, doesn't mean we should let it remain broken for the
> future.
> >> [MB] I'm not at all disagreeing on this point. Optimistically,
> I'd
> >> like
> >> to see the current implementations evolve to support this new
> >> approach
> >> as it will improve interoperability. If the WG can complete this
> >> work in
> >> a timely manner (perhaps the biggest issue given past
> performance),
> >> then
> >> the potential for uptake for a standardized implementation is
> INFO is
> >> far higher IMHO. If we can get this work chartered, then we're
> far
> >> more
> >> likely to get uptake by the other SDOs far sooner than if we
> >> continue to
> >> dillydally. And, it's important to remember that this isn't the
> only
> >> area where there are interoperability issues. And, actually, per
> >> SFSIW-1
> >> the use of INFO was not a common theme - either because it was a
> >> given
> >> or it's not as big an issue as some of us are assuming - in a
> quick
> >> scan
> >> I only found one document discussing the use of INFO.  [/MB]
> >> -Jonathan R.
> 
> _______________________________________________
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Cc: sip@ietf.org, "DOLLY, MARTIN C, ATTLABS" <mdolly@att.com>,
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Subject: Re: [Sip] INFO and what to do about it?
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Let me be clear: for the grandfathering part, I am suggesting that we 
develop a registry of *existing* usage, not that those existing usages 
change in any way. I want to get them written down, so at least there is 
a place to look to find out about something unknown to you, and to 
ensure nothing new gets defined that is in conflict with something old.

If those existing usages can be updated to use the new framework that 
would be better yet. But I'm not going to hold my breath waiting for 
that to happen.

	Paul

Michael Procter wrote:
> On Tue, June 24, 2008 3:00 pm, Paul Kyzivat wrote:
>> My latest thought is that we should "grandfather" existing uses of INFO.
>> We would provide a registry of them, without blessing them or
>> standardizing their specifications. That would at least shine some light
>> in the dark corners.
>>
>> At the same time, we would define the new INFO usage framework (still
>> TBD) and ban *new* INFO usages that don't follow it.
> 
> I agree.  The framework proposed in draft-kaplan-sip-info-events-01 seems
> like a step in the right direction.
> 
> And I agree with registering existing uses.   Even if existing uses aren't
> updated to use the negotiation mechanism suggested (or whatever solution
> the WG converges on), I think there is value in strongly encouraging the
> adoption of the 'Info-Package' header (or similar).  This shouldn't
> noticeably affect interop of legacy devices, but marking the requests like
> this would allow more recent devices to act appropriately in the presence
> of multiple incompatible non-standard extensions.  Not that such things
> exist, of course!
> 
> Regards,
> 
> Michael
> 
> 
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Christer Holmberg wrote:

>>> One option, as proposed by Paul, is to say that a UAC shall be able to RECEIVE reliable 199 responses.
>> Based on my comments above, I am reconsidering the advisability of that.
> 
> Well, I guess it wouldn't affect the complexness in the UAC, since it will have to be able to handle reliable provisional responses anyway (if it indicates support of it, that is).

Yes, I think it does increase the complexity in the implementation of 
199 handling.

You are right that the PRACK itself would be handled by the standard 
machinery for reliable provisionals. But then tearing down the dialog, 
which is what the 199 is supposed to trigger, can't be done upon receipt 
of a reliable 199. Rather it must wait until the after the PRACK is at 
least send, and perhaps until the 200 for the prack is received. And 
this is different than for the unreliable 199. This isn't rocket 
science, but it is more complexity. And of course it requires something 
that sends reliable 199s in order to test the UAC.

	Paul

> And, if the sender of the 199 is a B2BUA, I guess it COULD send it reliably.
> 
> But, I have no strong feelings regarding this.

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Cc: SIP@ietf.org, "DOLLY, MARTIN C, ATTLABS" <mdolly@att.com>,
	Mary Barnes <mary.barnes@nortel.com>,
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Subject: Re: [Sip] INFO and what to do about it?
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Robert,

I will agree that the scenario you pose is a possibility. And if it 
seems probable then we should do nothing.

The carrot we could offer would be the registry. New usages would get 
into the registry, but new usages of INFO without the framework would 
not be eligible for the registry.

Whether that is perceived as a carrot is an open question. It may be 
that most existing usages that haven't come through ietf don't *wish* to 
be public. If not, then there is nothing we can do.

	Paul

Robert Sparks wrote:
> inline
> 
> On Jun 24, 2008, at 9:00 AM, Paul Kyzivat wrote:
> 
>> Mary,
>>
>> My latest thought is that we should "grandfather" existing uses of 
>> INFO. We would provide a registry of them, without blessing them or 
>> standardizing their specifications. That would at least shine some 
>> light in the dark corners.
>>
>> At the same time, we would define the new INFO usage framework (still 
>> TBD) and ban *new* INFO usages that don't follow it.
> 
> Well, if we don't think banning INFO as a whole would be listened to, 
> why do we think banning INFO that doesn't use this new framework would 
> be listened to?
> 
> That's where I'm having difficulty with the question. I can see the bite 
> of work it would take to make the framework.
> I'm still trying to see what would actually use it if we did?
> Can somebody list the packages that we've already identified that we 
> would put out with this framework?
> 
>>
>>
>> I think that would be better than the current situation, and about as 
>> good as we can expect to achieve.
> 
> I'm still not convinced it makes it better. One outcome I can see from 
> following this path is two years from now finding ourselves with no 
> actual packages using the framework and  a set of _new_ INFO based 
> applications in the field that _don't_ use the framework.
> 
> Will we continue to put those in this proposed registry above? Or do we 
> just have another set of practices that we have to treat the current 
> non-standardized INFO uses with now?
> 
> The bet we would be placing by putting the work into creating this 
> framework is that it actually takes some motivation away from using a 
> non-standard bare INFO to realize some new application (and that there 
> are new applications out there where this would be a really good tool to 
> have to realize them). Is this framework really lowering a barrier to 
> entry or relieving some other pain so that the people who would 
> otherwise make a new non-standard use of INFO come use it? I'm having a 
> really hard time seeing that.
> 
> If we end up on the wrong side of that bet, we've arguably made things 
> _worse_ taking work away from other things, creating mechanics that have 
> to be implemented, tested and will ultimately be abused, and not solving 
> the actual problem in the first place.
> 
> So, I'll continue to look for some evidence this will actually get used 
> and relieves some pressure to just go use INFO in a proprietary way and 
> skip all this standards pain (hopefully folks here can help me with this).
> 
> Without it, I think the better path would be to create the registry you 
> describe, publish an informational document pointing out the pitfalls of 
> the approaches they take, and get on with other work.
> 
> I'll also, of course, accept a community decision to just do this work 
> and take the risk of losing the bet and help make sure we build the best 
> framework possible if that's where we really want to go. I just hope 
> somebody can convince me we're not trading off other work we could be 
> benefiting from more to find ourselves in effectively the same position 
> two years from now.
> 
> RjS
> 
> 
>>
>>
>>     Thanks,
>>     Paul
>>
>> Mary Barnes wrote:
>>> Just a couple more points of clarification on my view embedded below
>>> [MB].
>>> Mary
>>> -----Original Message-----
>>> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com] Sent: Tuesday, 
>>> June 24, 2008 8:25 AM
>>> To: Barnes, Mary (RICH2:AR00)
>>> Cc: Christer Holmberg; Robert Sparks; Paul Kyzivat; SIP@ietf.org; DOLLY,
>>> MARTIN C, ATTLABS
>>> Subject: Re: [Sip] INFO and what to do about it?
>>> Mary Barnes wrote:
>>>> I don't believe we could ever forbid INFO. I initially did not think 
>>>> we could accomplish anything around INFO, but I believe some of the 
>>>> work that's on the table would be useful for working towards 
>>>> interoperabilty for the INFO usages. I would be afraid to ask honestly
>>>> for the identification of all the different uses of INFO that are 
>>>> out there right now.
>>> I don't think we should be afraid of this at all.
>>> [MB] My guess (based on what I've seen) is that a lot of vendors would
>>> have around a half dozen (+- 2). Now, some of those might overlap given
>>> there is some level of interop between various vendors. [/MB]
>>> There are (sometimes/often) good reasons why folks resort to these
>>> solutions. Our job here at IETF is to ensure interoperability for the
>>> SIP protocol. If we don't listen to our customers - the people who have
>>> deployed and are actually using it - what purpose does our work serve?
>>> [MB] My experience is that at least half the uses of INFO were in place
>>> when the protocol was quite immature (i.e., well before RFC 3261). And,
>>> we indeed should listen to our customers by providing a flexible
>>> platform for them to do the things they need to do. IMHO, it's been loud
>>> and clear in the past that they want to use INFO and in the past, docs
>>> like "Info considered harmful" weren't helpful towards this end. [/MB]
>>>> Doing something is better than nothing at this point IMHO and I'm 
>>>> personally really tired of revisiting this issue every couple of 
>>>> years. AND, this would help us put a stake in the group on the 
>>>> future usages of INFO (whether we ever get rid of the old usages or 
>>>> not), as I believe there are other SDOs defining new uses of INFO 
>>>> right now to add to the mix of un-interoperability in this area.
>>> As long as SIP usage continues to rise, I suspect we will continue to
>>> see more INFO usages. Just because we cannot fix what is broken in the
>>> past, doesn't mean we should let it remain broken for the future.
>>> [MB] I'm not at all disagreeing on this point. Optimistically, I'd like
>>> to see the current implementations evolve to support this new approach
>>> as it will improve interoperability. If the WG can complete this work in
>>> a timely manner (perhaps the biggest issue given past performance), then
>>> the potential for uptake for a standardized implementation is INFO is
>>> far higher IMHO. If we can get this work chartered, then we're far more
>>> likely to get uptake by the other SDOs far sooner than if we continue to
>>> dillydally. And, it's important to remember that this isn't the only
>>> area where there are interoperability issues. And, actually, per SFSIW-1
>>> the use of INFO was not a common theme - either because it was a given
>>> or it's not as big an issue as some of us are assuming - in a quick scan
>>> I only found one document discussing the use of INFO.  [/MB]
>>> -Jonathan R.
> 
> 
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Subject: Re: [Sip] INFO and what to do about it?
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Henry,

You seem to be changing philosophy here. These existing uses of INFO are 
*running code*. Why doesn't that make them the good candidates for 
standardization?

	Paul

Henry Sinnreich wrote:
>> If we don't listen to our customers - the people who have
>> deployed and are actually using it - what purpose does our work serve?
> 
> This is what the managers of the typewriter business must have said when the
> word processors appeared: "Our customers ask for a white-out ribbon and for
> a price under $500."
> 
> A PC at that time cost $5,000 and the additional laser printer was another
> $5,000.
> 
> Some companies opted for the PC business nevertheless and did NOT listen to
> their customers.
> 
> Does anyone care for simple SIP that can be understood by developers with
> limited time on their hands?
> 
> Henry
> 
> 
> On 6/24/08 6:25 AM, "Jonathan Rosenberg" <jdrosen@cisco.com> wrote:
> 
>>
>> Mary Barnes wrote:
>>> I don't believe we could ever forbid INFO. I initially did not think
>>> we could accomplish anything around INFO, but I believe some of the
>>> work that's on the table would be useful for working towards
>>> interoperabilty for the INFO usages. I would be afraid to ask
>>> honestly for the identification of all the different uses of INFO
>>> that are out there right now.
>> I don't think we should be afraid of this at all.
>>
>> There are (sometimes/often) good reasons why folks resort to these
>> solutions. Our job here at IETF is to ensure interoperability for the
>> SIP protocol. If we don't listen to our customers - the people who have
>> deployed and are actually using it - what purpose does our work serve?
>>
>>
>>> Doing something is better than nothing at this point IMHO and I'm
>>> personally really tired of revisiting this issue every couple of
>>> years. AND, this would help us put a stake in the group on the future
>>> usages of INFO (whether we ever get rid of the old usages or not), as
>>> I believe there are other SDOs defining new uses of INFO right now to
>>> add to the mix of un-interoperability in this area.
>> As long as SIP usage continues to rise, I suspect we will continue to
>> see more INFO usages. Just because we cannot fix what is broken in the
>> past, doesn't mean we should let it remain broken for the future.
>>
>> -Jonathan R.
>>
> 
> 
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From: Jeroen van Bemmel <jbemmel@zonnet.nl>
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To: "Rockson Li (zhengyli)" <zhengyli@cisco.com>
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Cc: sip@ietf.org, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
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Christer,

My point was to formalize the moment a 199 is sent by relating it to the =

RFC3261 ICT state machine (BTW we're all silently assuming non-INVITE =

requests don't use provisional responses here)
So associated with PROCEEDING-->COMPLETED, but indeed only if a to-tag =

was received before (so not only 100)

Regards,
Jeroen

PS perhaps an idea to explicitly disallow 199 for non-INVITE requests?

Rockson Li (zhengyli) wrote:
> Yes, it's a typo, 100 does not establish early-dialog.
> -Rockson =

>
> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com] =

> Sent: Tuesday, June 24, 2008 6:32 PM
> To: Rockson Li (zhengyli); Jeroen van Bemmel
> Cc: sip@ietf.org
> Subject: RE: [Sip] Draft submission: draft-ietf-sip-199-00
>
>
> Hi, =

>
>   =

>>> Should the 199 response contain a Contact header? And if yes, in case =

>>> a proxy sends it, should it contain the address of that proxy (since th=
e UAS already sent a final response)?
>>>       =

>> IF the 199 is sent reliably be the proxy must contain a Contact header c=
ontaining the address of the proxy, yes.
>>
>>     =

>>> Should we say that a proxy may only generate and send a 199 when it =

>>> receives a final error response on an INVITE client Transaction which =

>>> was in the PROCEEDING state? (i.e. 1xx response was received before, =

>>> so conceptually sending the 199 response is an action associated with =

>>> the transition from PROCEEDING to COMPLETED)
>>>       =

>> I am not sure I understand. The idea IS to send 199 when a final error =

>> response is received by the forking proxy, if a 18x has previously been =
received.
>> [Rockson] PROCEEDING does not mean early-dialog is established, 100 =

>> Trying also move  INVITE client Transaction to PROCEEDING state.
>>     =

>
> [Christer] 199 is only sent when an early dialog has been established. 10=
0 Trying does not establish an early dialog.
>
> Regards,
>
> Christer
>
>
>
>
>
>
> Also, a forking proxy with multiple INVITE client Transaction may receive=
/forward 180 from one of them, and receive no provisional resp and final re=
sp directly from the other one, so  INVITE client Transaction's state is no=
t dependable. The decision making must be done in TU.
>
>
>
>
>
> Christer Holmberg wrote:
>   =

>> Hi,
>>
>> I agree it may be a good idea to not forbid sending it reliably.
>>
>> I do think it would be good to have text, saying that it can be sent unr=
eliable even if reliable responses are required, though, so that proxies ar=
en't forced to terminate PRACKs etc.
>>
>> Regards,
>>
>> Christer
>>  =

>>
>> -----Original Message-----
>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =

>> Dale.Worley@comcast.net
>> Sent: 21. kes=E4kuuta 2008 1:38
>> To: sip@ietf.org
>> Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
>>
>>    From: "Christer Holmberg" <christer.holmberg@ericsson.com>
>>
>>    [CHH] Whether the text should be in the document at all depends on if=
 we
>>    allow 199 to be sent reliably in the first place. Based on the commen=
ts
>>    received so far we should not mandate 199 to be sent reliably, even if
>>    100rel is required by the UAC. But, the question if whether we want to
>>    FORBID sending it reliably.
>>
>> If we ever might allow 199 to be used for HERFP, we should admit the pos=
sibility of sending it reliably in the first draft.  Otherwise, we'll be lo=
cked out of sending it reliably in the future.
>>
>> Dale
>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use =

>> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use =

>> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

>>     =

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

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


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From: "Dan Wing" <dwing@cisco.com>
To: "'Hannes Tschofenig'" <Hannes.Tschofenig@gmx.net>
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Cc: sip@ietf.org, 'Paul Kyzivat' <pkyzivat@cisco.com>,
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Subject: Re: [Sip] Toward the Evolution of SIP and Related Working Groups
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To summarize your email:  anybody that needs SIP security will use TLS between
their own proxies.  That does seem to be the consensus.  Perhaps how that
works should be written up -- as in, does that mean when I have a TLS
connection with boeing.com, I should only allow or only expect From: addresses
that end in @boeing.com, and not @big-airplane.boeing.com and not
@rolls-royce.com?

-d
 

> -----Original Message-----
> From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net] 
> Sent: Tuesday, June 24, 2008 1:17 AM
> To: Dan Wing
> Cc: 'Dean Willis'; 'Hadriel Kaplan'; sip@ietf.org; 'Paul Kyzivat'
> Subject: Re: [Sip] Toward the Evolution of SIP and Related 
> Working Groups
> 
> Many of the SIP security mechanisms share the same fate: They are far 
> ahead of the actual SIP deployment. This is true for SIP 
> Identity, SIP 
> CERT, SIP SAML, End-to-End Security, etc. When we started the 
> SAML work 
> we looked at what was going on at that time in the HTTP 
> space. Without 
> doubt the entire application layer identity management space 
> found a lot 
> of excitement. There is a lot of standardization being done 
> and also a 
> lot of deployment taking place. With the SIP space that was 
> obviously a 
> bit different and deployments today focus largely on voice (and there 
> not even on end-to-end SIP-based communication).
> 
> For example: Look at what is being used in XMPP. There is no 
> equalivalent of SIP Identity -- folks are currently looking into 
> providing certificates for server-to-server communication.
> 
> Even though they are, from a deployment point of view, ahead they are 
> not even close to where we are with our documents.
> 
> Ciao
> Hannes
> 
> PS: I also believe that the SIP Identity case isn't an easy 
> one either. 
> The guys that would make use of SIP Identity for a deployment 
> where the 
> two SIP proxies talk to each other there would not be a need for SIP 
> Identity since you are essentially replicating what TLS 
> provides you at 
> a lower layer already. In cases where there many SIP proxies 
> along the 
> path SIP Identity would be useful since it provides 
> protection against 
> any one of them being malicious. However, the guys who favour such a 
> deployment model are the onces that believe very much in the chain of 
> trust (hop-by-hop security). They have no interest in using 
> SIP Identity.
> 
> 
> 
> Dan Wing wrote:
> >>> draft-ietf-sip-saml
> >>>       
> >> -- I'm not sure this one is ever getting done.
> >>
> >> So I'm not sure there's enough there to justify a WG.
> >>
> >> How about an "Identity in SIP" working group that takes on 
> >> fixing RFC 4474 for gateways/b2buas and possibly considers 
> >> identity/role assertion using SAML?
> >>     
> >
> > I agree we need that.  I have tried, and so far failed, to
> > initiate activity towards such an effort.
> >
> > -d
> >
> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >   
> 

_______________________________________________
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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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Cc: sip@ietf.org, "DOLLY, MARTIN C, ATTLABS" <mdolly@att.com>,
	Mary Barnes <mary.barnes@nortel.com>,
	Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Sip] INFO and what to do about it?
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Paul Kyzivat wrote:
> Let me be clear: for the grandfathering part, I am suggesting that we
> develop a registry of *existing* usage, not that those existing usages
> change in any way. I want to get them written down, so at least there
> is a place to look to find out about something unknown to you, and to
> ensure nothing new gets defined that is in conflict with something
> old.
>
> If those existing usages can be updated to use the new framework that
> would be better yet. But I'm not going to hold my breath waiting for
> that to happen.

Exactly.  I agree with all this, and recognise it as a good start.

I went on to propose a half-way house between no changes for legacy
uses, and full adoption of a new framework.  A lightweight[0] framework
may[1] attract new uses, but the negotiation features present in
Hadriel's current draft seem likely to deter the migration of legacy
uses.  I am merely suggesting that using the INFO-labelling part of
Hadriel's draft but not the negotiation aspect could seem more
attractive[2] to legacy uses, and would still provide the rest of the
community some value (namely being able to identify all the INFO
messages flying around their networks).

Michael

[0] Both the implementation and documentation needs to be lightweight 
IMO.
[1] RjS' email discusses this further, and he seems less optimistic.
[2] Low implementation cost, relatively low risk, no additional stateful
behaviour required.

>
> Paul
>
> Michael Procter wrote:
>> On Tue, June 24, 2008 3:00 pm, Paul Kyzivat wrote:
>>> My latest thought is that we should "grandfather" existing uses of
>>> INFO.
>>> We would provide a registry of them, without blessing them or
>>> standardizing their specifications. That would at least shine some
>>> light
>>> in the dark corners.
>>>
>>> At the same time, we would define the new INFO usage framework
>>> (still
>>> TBD) and ban *new* INFO usages that don't follow it.
>>
>> I agree.  The framework proposed in draft-kaplan-sip-info-events-01
>> seems
>> like a step in the right direction.
>>
>> And I agree with registering existing uses.   Even if existing uses
>> aren't
>> updated to use the negotiation mechanism suggested (or whatever
>> solution
>> the WG converges on), I think there is value in strongly encouraging
>> the
>> adoption of the 'Info-Package' header (or similar).  This shouldn't
>> noticeably affect interop of legacy devices, but marking the requests
>> like
>> this would allow more recent devices to act appropriately in the
>> presence
>> of multiple incompatible non-standard extensions.  Not that such
>> things
>> exist, of course!
>>
>> Regards,
>>
>> Michael
>>
>>
>
>

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From sip-bounces@ietf.org  Tue Jun 24 12:13:19 2008
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Subject: Re: [Sip] INFO and what to do about it?
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On Jun 24, 2008, at 11:52 AM, Robert Sparks wrote:
>
> The bet we would be placing by putting the work into creating this  
> framework is that it actually takes some motivation away from using  
> a non-standard bare INFO to realize some new application (and that  
> there are new applications out there where this would be a really  
> good tool to have to realize them). Is this framework really  
> lowering a barrier to entry or relieving some other pain so that the  
> people who would otherwise make a new non-standard use of INFO come  
> use it? I'm having a really hard time seeing that.
>

Well, that's the whole reason we write RFCs instead of just telling  
people to innovate, isn't it?

> If we end up on the wrong side of that bet, we've arguably made  
> things _worse_ taking work away from other things, creating  
> mechanics that have to be implemented, tested and will ultimately be  
> abused, and not solving the actual problem in the first place.
>
> So, I'll continue to look for some evidence this will actually get  
> used and relieves some pressure to just go use INFO in a proprietary  
> way and skip all this standards pain (hopefully folks here can help  
> me with this).
>
> Without it, I think the better path would be to create the registry  
> you describe, publish an informational document pointing out the  
> pitfalls of the approaches they take, and get on with other work.

Don't use INFO. By the way, here's the "standard required" process for  
registering your non-standard INFO use by MIME-type and content- 
disposition. . . Don't blame us when it kills you, or you have to  
register a new MIME or content-disposition because somebody already  
signed up for the one you want to use.

I kind of like that approach. Can we do the same thing with P-headers?  
It'd sure save us a lot of work.

--
Dean
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From sip-bounces@ietf.org  Tue Jun 24 12:20:05 2008
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Subject: [Sip] Let's talk about a Dublin agenda
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I've been tipped off that we may be down to as little as two hours (or  
maybe 2.5)  of meeting time in Dublin due to additional WG and BOF  
events that didn't make it onto the earlier published agenda (like  
MEDIACTRL).

What are the absolutely most important things we need to talk about?  
What is the best possible use of our very valuable and very scarce  
time together?

I suspect that going by-rote through the list of open milestones is  
not the best use of our time.

Topics that I think we might want to talk about include but are not  
limited to:

Agenda bash and announcements usually use 10 minutes

1) Using SAML for SIP.
-- Milestone was Dec. 2007
-- Author has promised new rev before meeting
-- Is the WG committed to doing this work?
-- 10 minutes?

2) Test Matrix for SIP to Draft Standard
-- Milestone was Feb 2008
-- How are we going to move forward?
-- 10 minutes

3) Delivery of Request URI and Parameters to UAS Through Proxy
-- Milestone was Feb 2008
-- Conversation has lagged. Do we see a workable solution?
-- 30 minutes?

4) DTLS-SRTP
-- Milestone was Feb 2008
-- Delayed by the RFC 4474/Gateway/B2BUA
-- Decide: Do we hold DTLS-SRTP or publish with a caveat?
-- EKR may not be attending, can we resolve on-list? (no time?)

5) Mechanisms for UA Initiated Privacy
-- Milestone is July 2008
-- Author has promised new rev before meeting
-- 20 minutes?

6) Termination of early Dialog Prior to Final Response (199)
-- Milestone is November 2008
-- Substantial list activity
-- 30 minutes?

7) Guidelines for SIP Extension Modalities
-- No milestone
-- Needs conceptual discussion
-- Defer until #8 below is resolved? (no time?)

8) The INFO solution and Paul's compromise for historical registration
-- No milestone
-- Previous WG consensus to "do something"
-- 40 minutes?


Am I missing something that is critically important?

Which topics do we really need to talk about in order to move forward?
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From sip-bounces@ietf.org  Tue Jun 24 12:38:39 2008
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From: "Elwell, John" <john.elwell@siemens.com>
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Dean,

Well, just how exactly do we go about resolving the various Identify
issues (E.164 / SBCs / gateways)? I think Dan has tried unsuccessfully
for a BoF earlier. No agenda time in SIP this time. There have been some
huge email threads since Philadelphia. There are several relevant drafts
around (and possibly more before Dublin). 

John

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of Dean Willis
> Sent: 24 June 2008 20:20
> To: sip@ietf.org
> Cc: Keith Drage
> Subject: [Sip] Let's talk about a Dublin agenda
> 
> 
> I've been tipped off that we may be down to as little as two 
> hours (or  
> maybe 2.5)  of meeting time in Dublin due to additional WG and BOF  
> events that didn't make it onto the earlier published agenda (like  
> MEDIACTRL).
> 
> What are the absolutely most important things we need to talk about?  
> What is the best possible use of our very valuable and very scarce  
> time together?
> 
> I suspect that going by-rote through the list of open milestones is  
> not the best use of our time.
> 
> Topics that I think we might want to talk about include but are not  
> limited to:
> 
> Agenda bash and announcements usually use 10 minutes
> 
> 1) Using SAML for SIP.
> -- Milestone was Dec. 2007
> -- Author has promised new rev before meeting
> -- Is the WG committed to doing this work?
> -- 10 minutes?
> 
> 2) Test Matrix for SIP to Draft Standard
> -- Milestone was Feb 2008
> -- How are we going to move forward?
> -- 10 minutes
> 
> 3) Delivery of Request URI and Parameters to UAS Through Proxy
> -- Milestone was Feb 2008
> -- Conversation has lagged. Do we see a workable solution?
> -- 30 minutes?
> 
> 4) DTLS-SRTP
> -- Milestone was Feb 2008
> -- Delayed by the RFC 4474/Gateway/B2BUA
> -- Decide: Do we hold DTLS-SRTP or publish with a caveat?
> -- EKR may not be attending, can we resolve on-list? (no time?)
> 
> 5) Mechanisms for UA Initiated Privacy
> -- Milestone is July 2008
> -- Author has promised new rev before meeting
> -- 20 minutes?
> 
> 6) Termination of early Dialog Prior to Final Response (199)
> -- Milestone is November 2008
> -- Substantial list activity
> -- 30 minutes?
> 
> 7) Guidelines for SIP Extension Modalities
> -- No milestone
> -- Needs conceptual discussion
> -- Defer until #8 below is resolved? (no time?)
> 
> 8) The INFO solution and Paul's compromise for historical registration
> -- No milestone
> -- Previous WG consensus to "do something"
> -- 40 minutes?
> 
> 
> Am I missing something that is critically important?
> 
> Which topics do we really need to talk about in order to move forward?
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
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Use sipping@ietf.org for new developments on the application of sip


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Subject: Re: [Sip] Toward the Evolution of SIP and Related Working Groups
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Dan Wing wrote:
> To summarize your email:  anybody that needs SIP security will use
> TLS between their own proxies.  That does seem to be the consensus.
> Perhaps how that works should be written up -- as in, does that mean
> when I have a TLS connection with boeing.com, I should only allow or
> only expect From: addresses that end in @boeing.com, and not
> @big-airplane.boeing.com and not @rolls-royce.com?

Dan: I suspect you are talking of the above in the context
of SIP SAML.  Otherwise, for certificate-based authentication
between proxies, some of what you write above is discussed in
the sip-domain-certs draft.

Thanks,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
2701 Lucent Lane, Rm. 9F-546, Lisle, Illinois 60532 (USA)
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From: Dean Willis <dean.willis@softarmor.com>
To: "Elwell, John" <john.elwell@siemens.com>
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Subject: Re: [Sip] Let's talk about a Dublin agenda
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On Jun 24, 2008, at 2:38 PM, Elwell, John wrote:

> Dean,
>
> Well, just how exactly do we go about resolving the various Identify
> issues (E.164 / SBCs / gateways)? I think Dan has tried unsuccessfully
> for a BoF earlier. No agenda time in SIP this time. There have been  
> some
> huge email threads since Philadelphia. There are several relevant  
> drafts
> around (and possibly more before Dublin).
>

My suggestion is in two parts:

1) For DTLS-SRTP Framework, simply document the limitation of RFC 4474  
and move on. This is a one-paragraph add to the framework.

2) For the remainder, I'd like to suggest the "REal Time Identity and  
Role Expression" (RETIRE) Working Group, which would take over the RFC  
4474 revision along with SIP-SAML.

--
Dean

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From: Eric Rescorla <ekr@networkresonance.com>
To: Dean Willis <dean.willis@softarmor.com>
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Subject: Re: [Sip] A proposal for breaking the DTLS-SRTP vs RFC4474
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Dean,

This sounds like an excellent way to proceed. I'll generate some candidate text
and send it to the list in the next few days.

-Ekr

On Tue, Jun 24, 2008 at 10:21 AM, Dean Willis <dean.willis@softarmor.com> wrote:
>
> We've gotten stuck on a fine point in DTLS-SRTP.
>
> The current draft-ietf-sip-dtls-srtp-framework-01 uses an RFC 4474 Identity
> header to preserve the integrity of the media key's fingerprint, thereby
> detecting a certain class of MITM attack.
>
> However, RFC 4474 Identity headers are of questionable validity when used
> with protocol gateways or B2BUAs.  More or less, they're capable of
> asserting the identity of the gateway, not the identity of the calling
> party. But the recipient has no real way to figure out which is which.
>
> We've debated at some length, and with no good result, about whether we
> should try and fix RFC 4474. We've had some suggestions that may work for
> B2BUAs, and some other suggestions that may work for gateways, but we
> certainly don't have a consensus.
>
> That leaves our chartered deliverable of DTLS-SRTP hanging, and the
> milestone has gone past months ago.
>
> Here's a proposal:
>
> We add a caveat about the limitation of RFC 4474 to
> draft-ietf-sip-dtls-srtp-framework and go ahead and advance that
> specification. If somebody later decides to fix RFC 4474, they can do so,
> and if necessary update DTLS-SRTP if needed.
>
>
> Does that work for everybody?
>
> If we agree to it, I suggest that we move the date for WGLC of
> draft-ietf-sip-dtls-srtp-framework to July 2008, and move the milestone for
> delivery of that doc to the IESG into September.
>
> --
> Dean
>

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To: "Dean Willis" <dean.willis@softarmor.com>, sip@ietf.org
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Subject: Re: [Sip] Let's talk about a Dublin agenda
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Hey - not our fault - we got our request in on time. 

That said, I would offer #2 is possibly the most important work item, as it will drive a coherent, scoped work plan. 

How is this for a far-fetched thought: SIP to DS is the *only* work item for the SIP work group. If we could not get X done in 10 years (yes, it has been that long for some of these things), let us charter a new work group to work on X.
Sent from my Verizon Wireless BlackBerry

-----Original Message-----
From: Dean Willis <dean.willis@softarmor.com>

Date: Tue, 24 Jun 2008 14:19:54 
To:sip@ietf.org
Cc:Keith Drage <drage@alcatel-lucent.com>
Subject: [Sip] Let's talk about a Dublin agenda



I've been tipped off that we may be down to as little as two hours (or  
maybe 2.5)  of meeting time in Dublin due to additional WG and BOF  
events that didn't make it onto the earlier published agenda (like  
MEDIACTRL).

What are the absolutely most important things we need to talk about?  
What is the best possible use of our very valuable and very scarce  
time together?

I suspect that going by-rote through the list of open milestones is  
not the best use of our time.

Topics that I think we might want to talk about include but are not  
limited to:

Agenda bash and announcements usually use 10 minutes

1) Using SAML for SIP.
-- Milestone was Dec. 2007
-- Author has promised new rev before meeting
-- Is the WG committed to doing this work?
-- 10 minutes?

2) Test Matrix for SIP to Draft Standard
-- Milestone was Feb 2008
-- How are we going to move forward?
-- 10 minutes

3) Delivery of Request URI and Parameters to UAS Through Proxy
-- Milestone was Feb 2008
-- Conversation has lagged. Do we see a workable solution?
-- 30 minutes?

4) DTLS-SRTP
-- Milestone was Feb 2008
-- Delayed by the RFC 4474/Gateway/B2BUA
-- Decide: Do we hold DTLS-SRTP or publish with a caveat?
-- EKR may not be attending, can we resolve on-list? (no time?)

5) Mechanisms for UA Initiated Privacy
-- Milestone is July 2008
-- Author has promised new rev before meeting
-- 20 minutes?

6) Termination of early Dialog Prior to Final Response (199)
-- Milestone is November 2008
-- Substantial list activity
-- 30 minutes?

7) Guidelines for SIP Extension Modalities
-- No milestone
-- Needs conceptual discussion
-- Defer until #8 below is resolved? (no time?)

8) The INFO solution and Paul's compromise for historical registration
-- No milestone
-- Previous WG consensus to "do something"
-- 40 minutes?


Am I missing something that is critically important?

Which topics do we really need to talk about in order to move forward?
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From: "Elwell, John" <john.elwell@siemens.com>
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Dean,

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com] 
> Sent: 24 June 2008 20:48
> To: Elwell, John
> Cc: sip@ietf.org; Keith Drage
> Subject: Re: [Sip] Let's talk about a Dublin agenda
> 
> 
> On Jun 24, 2008, at 2:38 PM, Elwell, John wrote:
> 
> > Dean,
> >
> > Well, just how exactly do we go about resolving the various Identify
> > issues (E.164 / SBCs / gateways)? I think Dan has tried 
> unsuccessfully
> > for a BoF earlier. No agenda time in SIP this time. There 
> have been  
> > some
> > huge email threads since Philadelphia. There are several relevant  
> > drafts
> > around (and possibly more before Dublin).
> >
> 
> My suggestion is in two parts:
> 
> 1) For DTLS-SRTP Framework, simply document the limitation of 
> RFC 4474  
> and move on. This is a one-paragraph add to the framework.
> 
> 2) For the remainder, I'd like to suggest the "REal Time 
> Identity and  
> Role Expression" (RETIRE) Working Group, which would take 
> over the RFC  
> 4474 revision along with SIP-SAML.
[JRE] Will that produce results before we all retire? But seriously, a
new WG might be a good idea, but how long to set it up? What needs to be
done at or before Dublin in that direction?

John 
> 
> --
> Dean
> 
> 
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To: "Dean Willis" <dean.willis@softarmor.com>,
	"Robert Sparks" <rjsparks@nostrum.com>
From: eburger@standardstrack.com
Date: Tue, 24 Jun 2008 20:20:52 +0000
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Subject: Re: [Sip] INFO and what to do about it?
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Here is my prognostication: if we document current usage, that is what will be used. I'm in a foul enough mood to suggest we document curent usage to foster some semblance of interoperation, do the honorable thing and note why life sucks, and then consider our work done. I would offer that any framework we come up would be nothing more than an academic exercise. I can already hear people say we need to publish how to interwork this new framework with Cisco INFO DTMF because, golly gee, that is the only thing their deployed equipment supports.
Sent from my Verizon Wireless BlackBerry

-----Original Message-----
From: Dean Willis <dean.willis@softarmor.com>

Date: Tue, 24 Jun 2008 14:12:49 
To:Robert Sparks <rjsparks@nostrum.com>
Cc:SIP@ietf.org, "DOLLY, MARTIN C, ATTLABS" <mdolly@att.com>,Paul Kyzivat <pkyzivat@cisco.com>, Mary Barnes <mary.barnes@nortel.com>,Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Sip] INFO and what to do about it?



On Jun 24, 2008, at 11:52 AM, Robert Sparks wrote:
>
> The bet we would be placing by putting the work into creating this  
> framework is that it actually takes some motivation away from using  
> a non-standard bare INFO to realize some new application (and that  
> there are new applications out there where this would be a really  
> good tool to have to realize them). Is this framework really  
> lowering a barrier to entry or relieving some other pain so that the  
> people who would otherwise make a new non-standard use of INFO come  
> use it? I'm having a really hard time seeing that.
>

Well, that's the whole reason we write RFCs instead of just telling  
people to innovate, isn't it?

> If we end up on the wrong side of that bet, we've arguably made  
> things _worse_ taking work away from other things, creating  
> mechanics that have to be implemented, tested and will ultimately be  
> abused, and not solving the actual problem in the first place.
>
> So, I'll continue to look for some evidence this will actually get  
> used and relieves some pressure to just go use INFO in a proprietary  
> way and skip all this standards pain (hopefully folks here can help  
> me with this).
>
> Without it, I think the better path would be to create the registry  
> you describe, publish an informational document pointing out the  
> pitfalls of the approaches they take, and get on with other work.

Don't use INFO. By the way, here's the "standard required" process for  
registering your non-standard INFO use by MIME-type and content- 
disposition. . . Don't blame us when it kills you, or you have to  
register a new MIME or content-disposition because somebody already  
signed up for the one you want to use.

I kind of like that approach. Can we do the same thing with P-headers?  
It'd sure save us a lot of work.

--
Dean
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From: "Elwell, John" <john.elwell@siemens.com>
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Eric,

I too think #2 is important, particularly if that is viewed as one means
of attacking the issues from SFSIW. I am thinking here that less is
more, and what can we throw out of the Draft Standard to reduce the
options, whilst at the same time retaining or adding whatever is needed
to give us the necessary resilience, security, NAT/firewall-tolerance,
etc..

John

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of eburger@standardstrack.com
> Sent: 24 June 2008 21:10
> To: Dean Willis; sip@ietf.org
> Subject: Re: [Sip] Let's talk about a Dublin agenda
> 
> Hey - not our fault - we got our request in on time. 
> 
> That said, I would offer #2 is possibly the most important 
> work item, as it will drive a coherent, scoped work plan. 
> 
> How is this for a far-fetched thought: SIP to DS is the 
> *only* work item for the SIP work group. If we could not get 
> X done in 10 years (yes, it has been that long for some of 
> these things), let us charter a new work group to work on X.
> Sent from my Verizon Wireless BlackBerry
> 
> -----Original Message-----
> From: Dean Willis <dean.willis@softarmor.com>
> 
> Date: Tue, 24 Jun 2008 14:19:54 
> To:sip@ietf.org
> Cc:Keith Drage <drage@alcatel-lucent.com>
> Subject: [Sip] Let's talk about a Dublin agenda
> 
> 
> 
> I've been tipped off that we may be down to as little as two 
> hours (or  
> maybe 2.5)  of meeting time in Dublin due to additional WG and BOF  
> events that didn't make it onto the earlier published agenda (like  
> MEDIACTRL).
> 
> What are the absolutely most important things we need to talk about?  
> What is the best possible use of our very valuable and very scarce  
> time together?
> 
> I suspect that going by-rote through the list of open milestones is  
> not the best use of our time.
> 
> Topics that I think we might want to talk about include but are not  
> limited to:
> 
> Agenda bash and announcements usually use 10 minutes
> 
> 1) Using SAML for SIP.
> -- Milestone was Dec. 2007
> -- Author has promised new rev before meeting
> -- Is the WG committed to doing this work?
> -- 10 minutes?
> 
> 2) Test Matrix for SIP to Draft Standard
> -- Milestone was Feb 2008
> -- How are we going to move forward?
> -- 10 minutes
> 
> 3) Delivery of Request URI and Parameters to UAS Through Proxy
> -- Milestone was Feb 2008
> -- Conversation has lagged. Do we see a workable solution?
> -- 30 minutes?
> 
> 4) DTLS-SRTP
> -- Milestone was Feb 2008
> -- Delayed by the RFC 4474/Gateway/B2BUA
> -- Decide: Do we hold DTLS-SRTP or publish with a caveat?
> -- EKR may not be attending, can we resolve on-list? (no time?)
> 
> 5) Mechanisms for UA Initiated Privacy
> -- Milestone is July 2008
> -- Author has promised new rev before meeting
> -- 20 minutes?
> 
> 6) Termination of early Dialog Prior to Final Response (199)
> -- Milestone is November 2008
> -- Substantial list activity
> -- 30 minutes?
> 
> 7) Guidelines for SIP Extension Modalities
> -- No milestone
> -- Needs conceptual discussion
> -- Defer until #8 below is resolved? (no time?)
> 
> 8) The INFO solution and Paul's compromise for historical registration
> -- No milestone
> -- Previous WG consensus to "do something"
> -- 40 minutes?
> 
> 
> Am I missing something that is critically important?
> 
> Which topics do we really need to talk about in order to move forward?
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> _______________________________________________
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> This list is for NEW development of the core SIP Protocol
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> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
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Oy vey.  That's less time than we had in Philly, and we got barely anything done there.

Why can't some RAI groups overlap - for example mediactrl or xcon at the same time as speermint or drinks, or let p2psip overlap with any of those 4?

-hadriel


> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Dean
> Willis
> Sent: Tuesday, June 24, 2008 3:20 PM
> To: sip@ietf.org
> Cc: Keith Drage
> Subject: [Sip] Let's talk about a Dublin agenda
>
>
> I've been tipped off that we may be down to as little as two hours (or
> maybe 2.5)  of meeting time in Dublin due to additional WG and BOF
> events that didn't make it onto the earlier published agenda (like
> MEDIACTRL).
>
> What are the absolutely most important things we need to talk about?
> What is the best possible use of our very valuable and very scarce
> time together?
>
> I suspect that going by-rote through the list of open milestones is
> not the best use of our time.
>
> Topics that I think we might want to talk about include but are not
> limited to:
>
> Agenda bash and announcements usually use 10 minutes
>
> 1) Using SAML for SIP.
> -- Milestone was Dec. 2007
> -- Author has promised new rev before meeting
> -- Is the WG committed to doing this work?
> -- 10 minutes?
>
> 2) Test Matrix for SIP to Draft Standard
> -- Milestone was Feb 2008
> -- How are we going to move forward?
> -- 10 minutes
>
> 3) Delivery of Request URI and Parameters to UAS Through Proxy
> -- Milestone was Feb 2008
> -- Conversation has lagged. Do we see a workable solution?
> -- 30 minutes?
>
> 4) DTLS-SRTP
> -- Milestone was Feb 2008
> -- Delayed by the RFC 4474/Gateway/B2BUA
> -- Decide: Do we hold DTLS-SRTP or publish with a caveat?
> -- EKR may not be attending, can we resolve on-list? (no time?)
>
> 5) Mechanisms for UA Initiated Privacy
> -- Milestone is July 2008
> -- Author has promised new rev before meeting
> -- 20 minutes?
>
> 6) Termination of early Dialog Prior to Final Response (199)
> -- Milestone is November 2008
> -- Substantial list activity
> -- 30 minutes?
>
> 7) Guidelines for SIP Extension Modalities
> -- No milestone
> -- Needs conceptual discussion
> -- Defer until #8 below is resolved? (no time?)
>
> 8) The INFO solution and Paul's compromise for historical registration
> -- No milestone
> -- Previous WG consensus to "do something"
> -- 40 minutes?
>
>
> Am I missing something that is critically important?
>
> Which topics do we really need to talk about in order to move forward?
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Michael Procter <michael@procter.org.uk>, Paul Kyzivat <pkyzivat@cisco.com>
Date: Tue, 24 Jun 2008 16:54:10 -0400
Thread-Topic: [Sip] INFO and what to do about it?
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Subject: Re: [Sip] INFO and what to do about it?
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> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Michael Procter
>
> I went on to propose a half-way house between no changes for legacy
> uses, and full adoption of a new framework.  A lightweight[0] framework
> may[1] attract new uses, but the negotiation features present in
> Hadriel's current draft seem likely to deter the migration of legacy
> uses.  I am merely suggesting that using the INFO-labelling part of
> Hadriel's draft but not the negotiation aspect could seem more
> attractive[2] to legacy uses, and would still provide the rest of the
> community some value (namely being able to identify all the INFO
> messages flying around their networks).

I don't think there's much value in documenting the labeling part without negotiation.  One of the main issues right now is you don't know which INFO mechanisms the far end wants/needs/supports.  There isn't really much issue with discerning what the INFO you received was for once you got it, 'cause the content-type has been unambiguous so far for all of the current usages I think (or majority thereof).  It was the potential for content-type ambiguity in the future that was one of the concerns of INFO, but the real driver for me at least is the "what does the other end want/expect" problem.

I'm all for making the "negotiation" part as trivial and KISS as possible - I started with what I thought was trivial, but people wanted to pile onto it to cover 3PCC and other scenarios.  I'm cool with going back to simple as a better-than-nothing approach. :)

-hadriel
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> > To summarize your email:  anybody that needs SIP security will use
> > TLS between their own proxies.  That does seem to be the consensus.
> > Perhaps how that works should be written up -- as in, does that mean
> > when I have a TLS connection with boeing.com, I should only allow or
> > only expect From: addresses that end in @boeing.com, and not
> > @big-airplane.boeing.com and not @rolls-royce.com?
> 
> Dan: I suspect you are talking of the above in the context
> of SIP SAML. 

That wasn't my intent.

> Otherwise, for certificate-based authentication
> between proxies, some of what you write above is discussed in
> the sip-domain-certs draft.

Yes, some of it is there.  Can that draft be extended to talk 
about validating the From of requests (and maybe responses, I'm
not sure) that come from over a TLS-authenticated connection?  Or
would that be out of scope for it?

-d


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Subject: Re: [Sip] INFO and what to do about it?
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Hadriel Kaplan wrote:
> I don't think there's much value in documenting the labeling part
> without negotiation.  One of the main issues right now is you don't
> know which INFO mechanisms the far end wants/needs/supports.  There
> isn't really much issue with discerning what the INFO you received was
> for once you got it, 'cause the content-type has been unambiguous so
> far for all of the current usages I think (or majority thereof).  It
> was the potential for content-type ambiguity in the future that was
> one of the concerns of INFO, but the real driver for me at least is
> the "what does the other end want/expect" problem.

That is exactly the use-case that I am concerned with.  I've seen more
than one 'application/xml' to date, and whilst I have so far been able
to disambiguate those based on content analysis, I fear more are on the
way.

But on reflection, I suppose that vendors unwilling to declare their
content more precisely are unlikely to add a new header instead.

> I'm all for making the "negotiation" part as trivial and KISS as
> possible - I started with what I thought was trivial, but people
> wanted
> to pile onto it to cover 3PCC and other scenarios.  I'm cool with
> going
> back to simple as a better-than-nothing approach. :)

I'll leave it in your capable hands then!

Regards,

Michael

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Subject: Re: [Sip] Toward the Evolution of SIP and Related Working Groups
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Dan Wing wrote:
> That wasn't my intent.

Dan: OK; sorry.

>> Otherwise, for certificate-based authentication
>> between proxies, some of what you write above is discussed in
>> the sip-domain-certs draft.
> 
> Yes, some of it is there.  Can that draft be extended to talk 
> about validating the From of requests (and maybe responses, I'm
> not sure) that come from over a TLS-authenticated connection?  Or
> would that be out of scope for it?

I believe that validating the From would be out of scope as
the draft currently stands.  The draft does discuss how a
receiving proxy (i.e., the one that did the passive accept)
may authenticate the sender (see S7.4,
http://tools.ietf.org/html/draft-ietf-sip-domain-certs-00#section-7.4),
but this discussion does not normatively involve the From
header.

Furthermore, trying to match sender identity as carried in
 From to the TLS-level connection identity as carried in the
certificate does not work well in the the hop-by-hop model.
For instance, the sender's From may as well be user@a.com, but
the last upstream proxy that opened up a connection to the
recipient could be some intermediary (proxy.com) not in the
a.com domain, thus making it impossible to enforce that the
 From must match the identity of the last upstream proxy in the
certificate.

Did you have some alternate or new thoughts on this to see if
we can hash out something that would still be in the context
of the domain-certs draft?  As it stands, Scott and I are about
to submit a final post WGLC revision on it by next week.

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}
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From: Henry Sinnreich <hsinnrei@adobe.com>
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> You seem to be changing philosophy here.

Not at all. The philosophy is very simple:

Use "simple SIP" - basically the rendezvous and session setup functions, be
they CS or P2P.

Everything else related to SIP applications and customer demands (I
understand Jonathan's desire) can be better accomplished with API's, SDKs,
IDE's and various tool kits as is customary in the industry.

(Unfortunately we have other tools: Concept papers, frameworks, etc.)

This presumes however respecting the e2e Internet principle, the "dumb
network" architecture and the applications in the endpoints.

Henry


On 6/24/08 11:05 AM, "Paul Kyzivat" <pkyzivat@cisco.com> wrote:

> Henry,
> 
> You seem to be changing philosophy here. These existing uses of INFO are
> *running code*. Why doesn't that make them the good candidates for
> standardization?
> 
> Paul
> 
> Henry Sinnreich wrote:
>>> If we don't listen to our customers - the people who have
>>> deployed and are actually using it - what purpose does our work serve?
>> 
>> This is what the managers of the typewriter business must have said when the
>> word processors appeared: "Our customers ask for a white-out ribbon and for
>> a price under $500."
>> 
>> A PC at that time cost $5,000 and the additional laser printer was another
>> $5,000.
>> 
>> Some companies opted for the PC business nevertheless and did NOT listen to
>> their customers.
>> 
>> Does anyone care for simple SIP that can be understood by developers with
>> limited time on their hands?
>> 
>> Henry
>> 
>> 
>> On 6/24/08 6:25 AM, "Jonathan Rosenberg" <jdrosen@cisco.com> wrote:
>> 
>>> 
>>> Mary Barnes wrote:
>>>> I don't believe we could ever forbid INFO. I initially did not think
>>>> we could accomplish anything around INFO, but I believe some of the
>>>> work that's on the table would be useful for working towards
>>>> interoperabilty for the INFO usages. I would be afraid to ask
>>>> honestly for the identification of all the different uses of INFO
>>>> that are out there right now.
>>> I don't think we should be afraid of this at all.
>>> 
>>> There are (sometimes/often) good reasons why folks resort to these
>>> solutions. Our job here at IETF is to ensure interoperability for the
>>> SIP protocol. If we don't listen to our customers - the people who have
>>> deployed and are actually using it - what purpose does our work serve?
>>> 
>>> 
>>>> Doing something is better than nothing at this point IMHO and I'm
>>>> personally really tired of revisiting this issue every couple of
>>>> years. AND, this would help us put a stake in the group on the future
>>>> usages of INFO (whether we ever get rid of the old usages or not), as
>>>> I believe there are other SDOs defining new uses of INFO right now to
>>>> add to the mix of un-interoperability in this area.
>>> As long as SIP usage continues to rise, I suspect we will continue to
>>> see more INFO usages. Just because we cannot fix what is broken in the
>>> past, doesn't mean we should let it remain broken for the future.
>>> 
>>> -Jonathan R.
>>> 
>> 
>> 

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From: "Dan Wing" <dwing@cisco.com>
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References: <E6C2E8958BA59A4FB960963D475F7AC30BE3C778B0@mail.acmepacket.com><408DFED6-302C-4525-BCD5-1E43B68369BE@softarmor.com><485FEDBA.9090304@cisco.com><549C8D7E-8F0D-43C8-B6C3-B61D5D0DE238@softarmor.com><E6C2E8958BA59A4FB960963D475F7AC30BE7D5C137@mail.acmepacket.com>	<5744EABA-0861-4B95-BD91-2C4F0AC92BFE@softarmor.com>	<08d401c8d57a$dc7216d0$c2f0200a@cisco.com>
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> -----Original Message-----
> From: Vijay K. Gurbani [mailto:vkg@alcatel-lucent.com] 
> Sent: Tuesday, June 24, 2008 2:30 PM
> To: Dan Wing
> Cc: sip@ietf.org
> Subject: Re: [Sip] Toward the Evolution of SIP and Related 
> Working Groups
> 
> Dan Wing wrote:
> > That wasn't my intent.
> 
> Dan: OK; sorry.

No worries.

> >> Otherwise, for certificate-based authentication
> >> between proxies, some of what you write above is discussed in
> >> the sip-domain-certs draft.
> > 
> > Yes, some of it is there.  Can that draft be extended to talk 
> > about validating the From of requests (and maybe responses, I'm
> > not sure) that come from over a TLS-authenticated connection?  Or
> > would that be out of scope for it?
> 
> I believe that validating the From would be out of scope as
> the draft currently stands.  The draft does discuss how a
> receiving proxy (i.e., the one that did the passive accept)
> may authenticate the sender (see S7.4,
> http://tools.ietf.org/html/draft-ietf-sip-domain-certs-00#sect
> ion-7.4),
> but this discussion does not normatively involve the From
> header.

Right.

> Furthermore, trying to match sender identity as carried in
> From to the TLS-level connection identity as carried in the
> certificate does not work well in the the hop-by-hop model.
> For instance, the sender's From may as well be user@a.com, but
> the last upstream proxy that opened up a connection to the
> recipient could be some intermediary (proxy.com) not in the
> a.com domain, thus making it impossible to enforce that the
> From must match the identity of the last upstream proxy in the
> certificate.

Agreed.

> Did you have some alternate or new thoughts on this to see if
> we can hash out something that would still be in the context
> of the domain-certs draft?  As it stands, Scott and I are about
> to submit a final post WGLC revision on it by next week.

I don't think extending domain-cert's scope would be wise --
rather, I do think this is a separate problem.  The separate
problem is that with *some domains* you connect to (and do
domain-certs validation with), you trust them to claim and
use any old "From" they want -- because with those domains
you trust them to have done some validation of their previous
hop, who validated their previous hop, etc.  This is how
you would configure a SIP trunk with a service provider, for
example.

However, if you are communicating directly with another 
organization then you would not want to allow them to assert
any identity they wished, because the only identity you expect 
them to send you requests with a From that had their own 
identity (@microsoft.com, @boeing.com, etc.) -- that is, 
the identity of their own employees.  You do not expect them 
to assert the identity belonging to some other company.  You 
would not extend the transitive trust to them.


I do wish there was more interest in cryptographic end-to-end 
identity that survives through B2BUAs operated by service providers.  
It is the end goal; service provider B2BUAs/SBCs are not going 
away!

-d

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Dan Wing wrote:
> However, if you are communicating directly with another 
> organization then you would not want to allow them to assert
> any identity they wished, because the only identity you expect 
> them to send you requests with a From that had their own 
> identity (@microsoft.com, @boeing.com, etc.) -- that is, 
> the identity of their own employees.  You do not expect them 
> to assert the identity belonging to some other company.  You 
> would not extend the transitive trust to them.
> 
> I do wish there was more interest in cryptographic end-to-end 
> identity that survives through B2BUAs operated by service providers.  
> It is the end goal; service provider B2BUAs/SBCs are not going 
> away!

The sipsec URI (http://tools.ietf.org/html/draft-gurbani-sip-sipsec-01).
Sorry could not resist ;-)

Though that is not a panacea either since I doubt any B2BUA
operated by the service provider will agree to behave as a
transparent bit forwarder (although there are crypto-
techniques to allow intermediaries to snoop in an encrypted
stream to look only for certain keywords.  But I doubt that work
is to a point that one can create a scalable production system
out of it.)

There is also some work on using IBE (see
http://tools.ietf.org/html/draft-kupwade-sip-iba-00); this draft
was discussed on the list before the PHL IETF.  IIRC, the
discussions centered around problems with key escrow.

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: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Hadriel Kaplan" <HKaplan@acmepacket.com>,
	"Elwell, John" <john.elwell@siemens.com>,
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Subject: Re: [Sip] Progress draft-holmberg-sip-keep
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Hi,

As I said before, I would be happy to extend the scope to also cover proxy-=
proxy. I have also received an off-line comment indicating that it would be=
 useful. I don't think it would change the mechanism itself, so...

So, if we choose to move forward with the draft I would then update the sco=
pe etc it in the next version.

Does that sound ok?

Regards,

Christer =


-----Original Message-----
From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com] =

Sent: 24. kes=E4kuuta 2008 19:11
To: Elwell, John; DRAGE, Keith (Keith); sip@ietf.org
Cc: Christer Holmberg
Subject: RE: [Sip] Progress draft-holmberg-sip-keep



Those cases are the ones I said would be useful in my original email far be=
low.  I.e., when I said "two reasons not cited: PBX connections a la SIP tr=
unks, and proxy-proxy connections."  They are not identical, in the sense t=
hat an IP-PBX can be a b2bua/UA/whatever, but also in the sense of their ex=
ample use-case, for example an IP-PBX could be behind a NAT and thus want t=
his keep-alive for more reasons than a proxy-proxy.

-hadriel

> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens.com]
> Sent: Tuesday, June 24, 2008 11:48 AM
> To: Hadriel Kaplan; DRAGE, Keith (Keith); sip@ietf.org
> Cc: Christer Holmberg
> Subject: RE: [Sip] Progress draft-holmberg-sip-keep
>
> Hadriel,
>
> That trunking case is no different from other trunking cases, e.g., =

> proxy-to-proxy, and I have already suggested to Christer that he might =

> extend the scope of the draft to cover that. It would be good to have =

> some discussion on whether keep-alive on "trunking" interfaces would =

> be beneficial - I have an open mind on that at present.
>
> John
>
> > -----Original Message-----
> > From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]
> > Sent: 24 June 2008 16:32
> > To: Elwell, John; DRAGE, Keith (Keith); sip@ietf.org
> > Cc: Christer Holmberg
> > Subject: RE: [Sip] Progress draft-holmberg-sip-keep
> >
> > Hey John,
> > For the case where the IP-PBX registers into the SP, outbound can be =

> > used.  But it's the other case I was talking about:
> > where the IP-PBX does NOT register into the SP, but it instead is a =

> > trunk.
> >
> > -hadriel
> >
> >
> > > -----Original Message-----
> > > From: Elwell, John [mailto:john.elwell@siemens.com]
> > > Sent: Tuesday, June 24, 2008 3:15 AM
> > > To: Hadriel Kaplan; DRAGE, Keith (Keith); sip@ietf.org
> > > Cc: Christer Holmberg
> > > Subject: RE: [Sip] Progress draft-holmberg-sip-keep
> > >
> > > Hadriel,
> > >
> > > Concerning your "PBX connections a la SIP trunks" use case, I am =

> > > not convinced of this. ETSI TISPAN has specified two ways for
> > an IPPBX to
> > > connect to a service provider. One is the so-called
> > subscription-based
> > > approach, where the IPPBX registers with the SP and
> > communicates via an
> > > edge proxy. In this case, why not use SIP-outbound? The other is =

> > > the so-called peering-based approach, which is essentially the
> > same as any
> > > SIP "trunk", e.g., proxy-to-proxy, B2BUA-to-B2BUA, proxy-to-gateway.
> > > SIP-outbound does not apply to these situations, and similarly the =

> > > keep-alive mechanism is not specified for this cases. The
> > requirements
> > > in SIP-keep do not cover these situations.
> > >
> > > Basically I am not sold on the idea of a separate SIP-keep spec - =

> > > I don't think it would be the best use of WG time.
> > >
> > > John
> > >
> > >
> > > > -----Original Message-----
> > > > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On =

> > > > Behalf Of Hadriel Kaplan
> > > > Sent: 20 June 2008 18:20
> > > > To: DRAGE, Keith (Keith); sip@ietf.org
> > > > Cc: Christer Holmberg
> > > > Subject: Re: [Sip] Progress draft-holmberg-sip-keep
> > > >
> > > >
> > > > My 2 cents: it is a useful draft.  Personally, I would like to =

> > > > have it available for two reasons not cited: PBX connections a =

> > > > la SIP trunks, and proxy-proxy connections.
> > > > Today I think a lot of people are using OPTIONS requests or =

> > > > proprietary means to perform such keep-alives when registration =

> > > > is not appropriate, but it has led to some interop issues and =

> > > > performance concerns in some cases.
> > > >
> > > > Since the contentious issue of what form the keep-alives take =

> > > > have already been agreed on for outbound, this seems like a =

> > > > simple draft to get done. (famous last words, I know)
> > > >
> > > > -hadriel
> > > >
> > > >
> > > > > -----Original Message-----
> > > > > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
> > > > Behalf Of
> > > > > DRAGE, Keith (Keith)
> > > > > Sent: Thursday, June 19, 2008 5:31 AM
> > > > > To: sip@ietf.org
> > > > > Cc: Christer Holmberg
> > > > > Subject: [Sip] Progress draft-holmberg-sip-keep
> > > > >
> > > > > (As SIP WG cochair)
> > > > >
> > > > > We have been asked by the author of
> > > > >
> > > > >
> > http://www.ietf.org/internet-drafts/draft-holmberg-sip-keep-01.txt
> > > > >
> > > > > Whether the SIP WG can progress this document.
> > > > >
> > > > > Because this draft arose as a result of the discussion of
> > > > outbound, and
> > > > > indeed seems to reuse the requirements from outbound, and =

> > > > > these requirements never really got handled in the SIPPING
> > WG, it has been
> > > > > agreed with the SIPPING chairs that we will handle this
> > > > entirely within
> > > > > SIP.
> > > > >
> > > > > Now in order to ask for charter milestones, and indeed when
> > > > we finally
> > > > > present this to IESG, we will be asked for the level of
> > > > support in the
> > > > > WG, which is also predicated on does this fix a real
> > > > problem, or is it
> > > > > just a corner case with limited application. So:
> > > > >
> > > > > QUESTION 1 TO SIP WG: Are the use cases sufficiently
> > important to
> > > > > proceed with this draft? The document states:
> > > > >
> > > > >    Chapter 3.5 of draft-ietf-sip-outbound-13
> > [I-D.ietf-sip-outbound]
> > > > >    defines two keep-alive techniques.  Even though the
> > keep-alive
> > > > >    techniques are separated from the Outbound mechanism
> > > > >    [I-D.ietf-sip-outbound], it is currently not
> > possible to indicate
> > > > >    support of the keep-alive techniques without also
> > > > indicating support
> > > > >    for the Outbound mechanism.
> > > > >
> > > > >    The Outbound mechanism is enabled during the UA
> > > > registration phase.
> > > > >    However, there are use-cases where the UA does not
> > > > register itself,
> > > > >    but still needs to be able to make calls and maintain
> > > > NAT bindings
> > > > >    open during the duration of that call.  A typical example is
> > > > >    emergency calls.  There are also cases where entities do
> > > > not support
> > > > >    the Outbound mechanism, but still want to be able to
> > > > indicate support
> > > > >    and use the keep-alive techniques defined in
> > > > [I-D.ietf-sip-outbound].
> > > > >
> > > > > At first sight this is not the most inspiring declaration
> > > > of the need
> > > > > for the document. Please respond indicating whether you
> > > > consider this a
> > > > > useful draft, and propose text that you think would be
> > > > useful in this
> > > > > section. Conversely, if you think this draft is not useful
> > > > and the WG
> > > > > has other more important things to work on first, please
> > > > also respond.
> > > > >
> > > > > QUESTION 2 TO SIP WG: Do we have a robust set of
> > requirements for
> > > > > proceeding with this work? The document currently lists:
> > > > >
> > > > >    REQ 1: It MUST be possible for a UA to indicate support
> > > > of the keep-
> > > > >    alive techniques defined [I-D.ietf-sip-outbound] if the
> > > > UA supports
> > > > >    only the keep-alive part of [I-D.ietf-sip-outbound].
> > > > >
> > > > >    REQ 2: It MUST be possible for an edge proxy to indicate
> > > > support of
> > > > >    the keep-alive techniques defined
> > > > [I-D.ietf-sip-outbound] if the edge
> > > > >    poxy supports only the keep-alive part of
> > > > [I-D.ietf-sip-outbound].
> > > > >
> > > > > It would be desirable to agree these at the outset, and not
> > > > revisit them
> > > > > if we continue with the work. So if you require clarification, =

> > > > > modification, or addition to these two requirements, then
> > > > please also
> > > > > response with your questions and proposals.
> > > > >
> > > > > I suggest we would like responses by 30th June 2008 in
> > > > order to allow
> > > > > the author to revise the document before the deadlines.
> > > > Please note that
> > > > > we are looking to make this decision on the list within
> > > > this deadline
> > > > > based on responses received, not leave it until the
> > Dublin meeting.
> > > > >
> > > > > Regards
> > > > >
> > > > > Keith
> > > > > _______________________________________________
> > > > > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > > > > This list is for NEW development of the core SIP Protocol Use =

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

> > > > sip-implementors@cs.columbia.edu for questions on current sip =

> > > > Use sipping@ietf.org for new developments on the
> > application of sip
> > > >
> >
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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> Dan Wing wrote:
> > However, if you are communicating directly with another 
> > organization then you would not want to allow them to assert
> > any identity they wished, because the only identity you expect 
> > them to send you requests with a From that had their own 
> > identity (@microsoft.com, @boeing.com, etc.) -- that is, 
> > the identity of their own employees.  You do not expect them 
> > to assert the identity belonging to some other company.  You 
> > would not extend the transitive trust to them.
> > 
> > I do wish there was more interest in cryptographic end-to-end 
> > identity that survives through B2BUAs operated by service 
> > providers.  
> > It is the end goal; service provider B2BUAs/SBCs are not going 
> > away!
> 
> The sipsec URI 
> (http://tools.ietf.org/html/draft-gurbani-sip-sipsec-01).
> Sorry could not resist ;-)

heh.  Yeah, I think SIP four-oh had a few ideas in that space,
too.

> Though that is not a panacea either since I doubt any B2BUA
> operated by the service provider will agree to behave as a
> transparent bit forwarder (although there are crypto-
> techniques to allow intermediaries to snoop in an encrypted
> stream to look only for certain keywords.  But I doubt that work
> is to a point that one can create a scalable production system
> out of it.)
> 
> There is also some work on using IBE (see
> http://tools.ietf.org/html/draft-kupwade-sip-iba-00); this draft
> was discussed on the list before the PHL IETF.  IIRC, the
> discussions centered around problems with key escrow.

There are also http://tools.ietf.org/html/draft-wing-sip-identity-media-02 and
http://tools.ietf.org/html/draft-fischer-sip-e2e-sec-media-00, both of which
propose slightly different techniques that are a middleground between sipsec
and (today's) blind transitive trust.

-d

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On Jun 24, 2008, at 3:17 PM, Elwell, John wrote:

>>
>>
>> 2) For the remainder, I'd like to suggest the "REal Time
>> Identity and
>> Role Expression" (RETIRE) Working Group, which would take
>> over the RFC
>> 4474 revision along with SIP-SAML.
> [JRE] Will that produce results before we all retire? But seriously, a
> new WG might be a good idea, but how long to set it up? What needs  
> to be
> done at or before Dublin in that direction?

Well, a new working group starts with a BOF request and a draft  
charter and somebody who wants the WG badly enough to do the leg work  
and chair the BOF.

It's a little late in the cycle to get one of those in for Dublin, but  
we could certainly start talking about a draft charter, then run a BOF  
at the next meeting. That would fit squarely with my goal of ramping  
down the "eternal SIP group" by fall 2009.

--
Dean
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On Jun 24, 2008, at 3:09 PM, eburger@standardstrack.com wrote:

> Hey - not our fault - we got our request in on time.
>
> That said, I would offer #2 is possibly the most important work  
> item, as it will drive a coherent, scoped work plan.


#2 was:

2) Test Matrix for SIP to Draft Standard
-- Milestone was Feb 2008
-- How are we going to move forward?
-- 10 minutes
>
>
>
> How is this for a far-fetched thought: SIP to DS is the *only* work  
> item for the SIP work group. If we could not get X done in 10 years  
> (yes, it has been that long for some of these things), let us  
> charter a new work group to work on X.
> Sent from my Verizon Wireless BlackBerry

How about SIP to Draft Standard as the only work item for a new WG?  
That's a pretty crisply defined charter. Our current milestone is only  
to define a test matrix; said test matrix being input to some other  
work process that either takes SIP to Draft or decides it can't be done.

--
Dean

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> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of Dean Willis
> Sent: Tuesday, June 24, 2008 12:48 PM
> To: Elwell, John
> Cc: sip@ietf.org; Keith Drage
> Subject: Re: [Sip] Let's talk about a Dublin agenda
> 
> 
> On Jun 24, 2008, at 2:38 PM, Elwell, John wrote:
> 
> > Dean,
> >
> > Well, just how exactly do we go about resolving the various Identify
> > issues (E.164 / SBCs / gateways)? I think Dan has tried 
> unsuccessfully
> > for a BoF earlier. No agenda time in SIP this time. There 
> have been  
> > some
> > huge email threads since Philadelphia. There are several relevant  
> > drafts
> > around (and possibly more before Dublin).
> >
> 
> My suggestion is in two parts:
> 
> 1) For DTLS-SRTP Framework, simply document the limitation of 
> RFC 4474  
> and move on. This is a one-paragraph add to the framework.

Sounds good.

> 2) For the remainder, I'd like to suggest the "REal Time 
> Identity and  
> Role Expression" (RETIRE) Working Group, which would take 
> over the RFC  
> 4474 revision along with SIP-SAML.

Sounds good, too.

-d

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Works for me.  Thanks,
-d


> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of Dean Willis
> Sent: Tuesday, June 24, 2008 10:22 AM
> To: sip@ietf.org; Eric Rescorla; Jason Fischl
> Cc: Cullen Jennings; Keith Drage
> Subject: [Sip] A proposal for breaking the DTLS-SRTP vs 
> RFC4474 gatewaydeadlock
> 
> 
> We've gotten stuck on a fine point in DTLS-SRTP.
> 
> The current draft-ietf-sip-dtls-srtp-framework-01 uses an RFC 4474  
> Identity header to preserve the integrity of the media key's  
> fingerprint, thereby detecting a certain class of MITM attack.
> 
> However, RFC 4474 Identity headers are of questionable validity when  
> used with protocol gateways or B2BUAs.  More or less, they're 
> capable  
> of asserting the identity of the gateway, not the identity of the  
> calling party. But the recipient has no real way to figure out which  
> is which.
> 
> We've debated at some length, and with no good result, about whether  
> we should try and fix RFC 4474. We've had some suggestions that may  
> work for B2BUAs, and some other suggestions that may work for  
> gateways, but we certainly don't have a consensus.
> 
> That leaves our chartered deliverable of DTLS-SRTP hanging, and the  
> milestone has gone past months ago.
> 
> Here's a proposal:
> 
> We add a caveat about the limitation of RFC 4474 to draft-ietf-sip- 
> dtls-srtp-framework and go ahead and advance that specification. If  
> somebody later decides to fix RFC 4474, they can do so, and if  
> necessary update DTLS-SRTP if needed.
> 
> 
> Does that work for everybody?
> 
> If we agree to it, I suggest that we move the date for WGLC of draft- 
> ietf-sip-dtls-srtp-framework to July 2008, and move the 
> milestone for  
> delivery of that doc to the IESG into September.
> 
> --
> Dean
> _______________________________________________
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Subject: Re: [Sip] Let's talk about a Dublin agenda
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> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of Dean Willis
> Sent: Tuesday, June 24, 2008 3:38 PM
> To: Elwell, John
> Cc: sip@ietf.org; Keith Drage
> Subject: Re: [Sip] Let's talk about a Dublin agenda
> 
> 
> On Jun 24, 2008, at 3:17 PM, Elwell, John wrote:
> 
> >>
> >>
> >> 2) For the remainder, I'd like to suggest the "REal Time
> >> Identity and
> >> Role Expression" (RETIRE) Working Group, which would take
> >> over the RFC
> >> 4474 revision along with SIP-SAML.
> > [JRE] Will that produce results before we all retire? But 
> seriously, a
> > new WG might be a good idea, but how long to set it up? What needs  
> > to be
> > done at or before Dublin in that direction?
> 
> Well, a new working group starts with a BOF request and a draft  
> charter and somebody who wants the WG badly enough to do the 
> leg work and chair the BOF.

You say that as if nobody tried before.  I requested a BoF for
Philadelphia, called Discussion and Analysis of SIP Identity 
(DASI).  It was turned down by the IESG, and did not have the
support of the RAI ADs.

I have received no indication that additional people in SIP cared 
about the problem (beyond the ~6 or so involved in the thread we
had on the list), nor an indication from the RAI ADs that our email 
threads on SIP after the Philadelphia meeting had swayed their 
opinion.

I could continue to beat my head against the wall, but I can't
see through the blood as it is.

> It's a little late in the cycle to get one of those in for 
> Dublin, but we could certainly start talking about a draft 
> charter, then run a BOF at the next meeting.  That would fit 
> squarely with my goal of ramping down the "eternal SIP group" 
> by fall 2009.

-d

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From: "Dan Wing" <dwing@cisco.com>
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	"'Dean Willis'" <dean.willis@softarmor.com>, <sip@ietf.org>
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Subject: Re: [Sip] Let's talk about a Dublin agenda
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Hadriel Kaplan wrote:

> Oy vey.  That's less time than we had in Philly, and we got 
> barely anything done there.
> 
> Why can't some RAI groups overlap - for example mediactrl or 
> xcon at the same time as speermint or drinks, or let p2psip 
> overlap with any of those 4?

I was shot down when I proposed it:
http://www.ietf.org/mail-archive/web/rai-discuss/current/threads.html#00009

I do hope we reconsider.

-d

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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Dan Wing" <dwing@cisco.com>, "Hadriel Kaplan" <HKaplan@acmepacket.com>,
	"Dean Willis" <dean.willis@softarmor.com>, <sip@ietf.org>
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Subject: Re: [Sip] Let's talk about a Dublin agenda
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Dan - your suggestion was for groups that have more overlap (e.g., SIP,
MMUSIC with XCON, MEDIACTRL).  The groups Hadriel is explicitly
suggesting might just work (well it does for me :). 

Mary.  
- who was the one that shot Dan down last time. 

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Dan Wing
Sent: Tuesday, June 24, 2008 6:07 PM
To: 'Hadriel Kaplan'; 'Dean Willis'; sip@ietf.org
Cc: 'Keith Drage'
Subject: Re: [Sip] Let's talk about a Dublin agenda

Hadriel Kaplan wrote:

> Oy vey.  That's less time than we had in Philly, and we got barely 
> anything done there.
> 
> Why can't some RAI groups overlap - for example mediactrl or xcon at 
> the same time as speermint or drinks, or let p2psip overlap with any 
> of those 4?

I was shot down when I proposed it:
http://www.ietf.org/mail-archive/web/rai-discuss/current/threads.html#00
009

I do hope we reconsider.

-d

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From: Dean Willis <dean.willis@softarmor.com>
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Subject: [Sip] First draft, RETIRE Charter
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This is a first-pass at what I think we might do with a RETIRE working  
group

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


The "REal Time Identity and  Role Expression" (RETIRE) Working Group's  
purpose is to standardize the expression and assertion of identity and  
roles needed for real-time communications with the IETF's Realtime  
Applications Infrastructure (RAI) area protocols, primarily the  
Session Initiation Protocol (SIP) and others used in conjunction with  
SIP.

By "Identity", we mean authenticated addresses of record (as defined  
in RFC 4474 "Enhancements for Authenticated Identity Management in the  
Session Initiation Protocol". RFC 4474 provides a means by which a SIP  
server may vouch for the "From" address of a request transiting that  
server. It further provides a cryptographic mechanism for testing the  
source of the assertion. In the typical case, this mechanism is  
dependent on the same Public Key Infrastructure that supports secure  
HTTP services. Since RFC 4474 provides only authentication of the  
source of SIP request messages and it is also important to determine  
the identity of the sender of responses, RFC 4916 "Connected Identity  
in the Session Initiation Protocol" provides authenticated identity in  
an ongoing SIP dialog by applying RFC 4474 to requests in the reverse  
direction. While adequate for some scenarios, this mechanism is not  
useful for SIP interactions having a single non-dialog transaction,  
such as MESSAGE requests. RFC 4474 suffers additional limitations  
relating to its use with gateways, session border controllers, and  
back-to-back user agents.

By "Role", we mean aspects of a a system user that are related  
identity, but more dynamic in nature. For example, Alice might  
delegate her calls to Alice. A calling party, expecting to reach Alice  
but reaching Bob, needs to know that this is a reasonable thing and  
what Bob's role in relation to Alice is. This problem is in general  
referred to as the "unanticipated respondent problem", and it has  
impacts both at the level of user expectations and key management.  
Proposals for fixing it have included deprecation of proxy retargeting  
and the use of a new SIP provisional response to inform the calling  
party about the retargeting operation.

Security Assertion Markup Language (SAML) is an XML-based standard for  
exchanging authentication and authorization data between security  
domains, that is, between an identity provider (a producer of  
assertions) and a service provider (a consumer of assertions). SAML is  
a product of the OASIS Security Services Technical Committee. The SIP  
working group worked for several years on applying SAML to SIP with  
mixed results, but the technique appears to have great promise.


The initial deliverables of the RETIRE working group are:

1) A standards-track revision of RFC 4474 that resolves its  
limitations with respect to gateways, session border controllers,  and  
back-to-back user agents. This revision will also address the use of  
Authenticated Identity Management in conjunction with SIP response  
messages.

2) Specify a solution for the Unanticipated Respondent Problem in SIP.

3) Guidelines for the use of SAML with SIP for the purposes of  
expressing identity and role, specifically in the contexts of #1 and  
#2 above.


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On Jun 24, 2008, at 5:55 PM, Dan Wing wrote:
>>>
>>
>> Well, a new working group starts with a BOF request and a draft
>> charter and somebody who wants the WG badly enough to do the
>> leg work and chair the BOF.
>
> You say that as if nobody tried before.  I requested a BoF for
> Philadelphia, called Discussion and Analysis of SIP Identity
> (DASI).  It was turned down by the IESG, and did not have the
> support of the RAI ADs.

True, However, the approach you took at that time wasn't clearly  
coordinated as a handoff of responsibility from the SIP working group.  
That's what I propose to try now.

We should also consider that much of the proposed work of RUCUS  
(Reducing Unwanted Communications Using SIP), aka anti-SPIT, depend on  
strong identity.

--
Dean

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On Wed, Jun 25, 2008 at 12:20 AM, Dean Willis <dean.willis@softarmor.com> wrote:

> For example, Alice might delegate her calls to Alice.

resulting in a horrible loop.

 ~ Theo
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That's because you posted that on a list no one pays attention to. :)
(who even knew there was an rai-discuss@ietf.org separate from rai@ietf.org??)

And you proposed overlapping things with SIP and MMUSIC, which does seem tough.
I'm thinking more like...

Thread-safe:
Thread-1          Thread-2        Thread-3
--------          --------        --------
speermint         xcon            p2psip
drinks            mediactrl
enum              speechsc


Global:
-------
avt
bliss
mmusic
simple
sip
sipping
ecrit
geopriv

Any Thread WG can conflict with a WG from another Thread.  No Global or Thread WG can conflict with a Global WG.
You could probably create a Thread-4 with ecrit and geopriv, but 3 concurrent threads seem enough.

-hadriel


> -----Original Message-----
> From: Dan Wing [mailto:dwing@cisco.com]
>
> I was shot down when I proposed it:
> http://www.ietf.org/mail-archive/web/rai-
> discuss/current/threads.html#00009
>
> I do hope we reconsider.
>
> -d

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> >> Well, a new working group starts with a BOF request and a draft
> >> charter and somebody who wants the WG badly enough to do the
> >> leg work and chair the BOF.
> >
> > You say that as if nobody tried before.  I requested a BoF for
> > Philadelphia, called Discussion and Analysis of SIP Identity
> > (DASI).  It was turned down by the IESG, and did not have the
> > support of the RAI ADs.
> 
> True, However, the approach you took at that time wasn't clearly  
> coordinated as a handoff of responsibility from the SIP 
> working group. 

True.

> That's what I propose to try now.

Ok.

> We should also consider that much of the proposed work of RUCUS  
> (Reducing Unwanted Communications Using SIP), aka anti-SPIT, 
> depend on strong identity.

Yes, it does.

-d

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From: "Dan Wing" <dwing@cisco.com>
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> That's because you posted that on a list no one pays attention to. :)
> (who even knew there was an rai-discuss@ietf.org separate 
> from rai@ietf.org??)

Neither are used much at all, for whatever that's worth.

> And you proposed overlapping things with SIP and MMUSIC, 
> which does seem tough.
> I'm thinking more like...
> 
> Thread-safe:
> Thread-1          Thread-2        Thread-3
> --------          --------        --------
> speermint         xcon            p2psip
> drinks            mediactrl
> enum              speechsc
> 
> 
> Global:
> -------
> avt
> bliss
> mmusic
> simple
> sip
> sipping
> ecrit
> geopriv
> 
> Any Thread WG can conflict with a WG from another Thread.  No 
> Global or Thread WG can conflict with a Global WG.
> You could probably create a Thread-4 with ecrit and geopriv, 
> but 3 concurrent threads seem enough.

Sounds great to me!

-d

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On Jun 24, 2008, at 4:50 PM, Dan Wing wrote:

>> That's because you posted that on a list no one pays attention to. :)
>> (who even knew there was an rai-discuss@ietf.org separate
>> from rai@ietf.org??)
>
> Neither are used much at all, for whatever that's worth.

I moved the main part of this thread over to rai@ietf. The rai-discuss  
list was formed very early before the area was formed and is supposed  
to be more or less dead. The right list for general rai area topics  
would be rai@ietf.org but as Dan points out, not much happens on it.

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On Jun 24, 2008, at 6:29 PM, Theo Zourzouvillys wrote:

> On Wed, Jun 25, 2008 at 12:20 AM, Dean Willis <dean.willis@softarmor.com 
> > wrote:
>
>> For example, Alice might delegate her calls to Alice.
>
> resulting in a horrible loop.


Well, that would be unanticipated! Same name, different person . . .

Thanks, Theo. I actually did mean "Alice might delegate her calls to  
Bob".

You win the actually-reading-what-I-wrote prize for the day.

--
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Cullen,

So is that a polite way of telling us that we should also join the rai@ietf.org list and post responses to today's thread about splitting up the SIP WG *there* versus here?

Dan (the other one)
-- 
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-----Original Message-----
From: Cullen Jennings <fluffy@cisco.com>

Date: Tue, 24 Jun 2008 19:48:58 
To:Dan Wing <dwing@cisco.com>
Cc:sip@ietf.org, 'Keith Drage' <drage@alcatel-lucent.com>,'Dean Willis' <dean.willis@softarmor.com>
Subject: Re: [Sip] Let's talk about a Dublin agenda



On Jun 24, 2008, at 4:50 PM, Dan Wing wrote:

>> That's because you posted that on a list no one pays attention to. :)
>> (who even knew there was an rai-discuss@ietf.org separate
>> from rai@ietf.org??)
>
> Neither are used much at all, for whatever that's worth.

I moved the main part of this thread over to rai@ietf. The rai-discuss
list was formed very early before the area was formed and is supposed
to be more or less dead. The right list for general rai area topics
would be rai@ietf.org but as Dan points out, not much happens on it.

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


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Subject: Re: [Sip] Let's talk about a Dublin agenda
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The discution of breaking out part of SIP to a new WG is probably best  
done on SIP list for now.

However, discussion of scheduling all the WG in RAI is probably best  
done on rai@ietf. Not a real big deal one way or the other.



On Jun 24, 2008, at 7:53 PM, Dan York wrote:

> Cullen,
>
> So is that a polite way of telling us that we should also join the rai@ietf.org 
>  list and post responses to today's thread about splitting up the  
> SIP WG *there* versus here?
>
> Dan (the other one)
> -- 
> Dan York, CISSP, Director of Emerging Communication Technology
> Office of the CTO    Voxeo Corporation     dyork@voxeo.com
> Phone: +1-407-455-5859  Skype: danyork  http://www.voxeo.com
> Blogs: http://blogs.voxeo.com  http://www.disruptivetelephony.com
>
> Bring your web applications to the phone.
> Find out how at http://evolution.voxeo.com
>
>
>
> -----Original Message-----
> From: Cullen Jennings <fluffy@cisco.com>
>
> Date: Tue, 24 Jun 2008 19:48:58
> To:Dan Wing <dwing@cisco.com>
> Cc:sip@ietf.org, 'Keith Drage' <drage@alcatel-lucent.com>,'Dean  
> Willis' <dean.willis@softarmor.com>
> Subject: Re: [Sip] Let's talk about a Dublin agenda
>
>
>
> On Jun 24, 2008, at 4:50 PM, Dan Wing wrote:
>
>>> That's because you posted that on a list no one pays attention  
>>> to. :)
>>> (who even knew there was an rai-discuss@ietf.org separate
>>> from rai@ietf.org??)
>>
>> Neither are used much at all, for whatever that's worth.
>
> I moved the main part of this thread over to rai@ietf. The rai-discuss
> list was formed very early before the area was formed and is supposed
> to be more or less dead. The right list for general rai area topics
> would be rai@ietf.org but as Dan points out, not much happens on it.
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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


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Date: Wed, 25 Jun 2008 08:24:17 +0100
From: "Elwell, John" <john.elwell@siemens.com>
In-reply-to: <F4A0F996-F7E2-4175-9EDE-8C169B167D40@softarmor.com>
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Dean,

Thanks for preparing this. I like what I see. This seems to be an ample
scope for a WG, and it could probably be addressed during a 1 to 1.5
hour slot at each IETF. A couple of comments below.

John


> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of Dean Willis
> Sent: 25 June 2008 00:20
> To: sip@ietf.org
> Subject: [Sip] First draft, RETIRE Charter
> 
> This is a first-pass at what I think we might do with a 
> RETIRE working  
> group
> 
> ---------------
> 
> 
> The "REal Time Identity and  Role Expression" (RETIRE) 
> Working Group's  
> purpose is to standardize the expression and assertion of 
> identity and  
> roles needed for real-time communications with the IETF's Realtime  
> Applications Infrastructure (RAI) area protocols, primarily the  
> Session Initiation Protocol (SIP) and others used in 
> conjunction with  
> SIP.
> 
> By "Identity", we mean authenticated addresses of record (as defined  
> in RFC 4474 "Enhancements for Authenticated Identity 
> Management in the  
> Session Initiation Protocol". RFC 4474 provides a means by 
> which a SIP  
> server may vouch for the "From" address of a request transiting that  
> server. It further provides a cryptographic mechanism for 
> testing the  
> source of the assertion. In the typical case, this mechanism is  
> dependent on the same Public Key Infrastructure that supports secure  
> HTTP services. Since RFC 4474 provides only authentication of the  
> source of SIP request messages and it is also important to determine  
> the identity of the sender of responses, RFC 4916 "Connected 
> Identity  
> in the Session Initiation Protocol" provides authenticated 
> identity in  
> an ongoing SIP dialog by applying RFC 4474 to requests in the 
> reverse  
> direction. While adequate for some scenarios, this mechanism is not  
> useful for SIP interactions having a single non-dialog transaction,  
> such as MESSAGE requests. RFC 4474 suffers additional limitations  
> relating to its use with gateways, session border controllers, and  
> back-to-back user agents.
[JRE] It also suffers an additional limitation to do with E.164 numbers.
So I would prefer to say:
"RFC 4474 suffers additional limitations relating to its use with E.164
numbers and with gateways, session border controllers, and back-to-back
user agents."

> 
> By "Role", we mean aspects of a a system user that are related  
> identity, but more dynamic in nature. For example, Alice might  
> delegate her calls to Alice. A calling party, expecting to 
> reach Alice  
> but reaching Bob, needs to know that this is a reasonable thing and  
> what Bob's role in relation to Alice is. This problem is in general  
> referred to as the "unanticipated respondent problem", and it has  
> impacts both at the level of user expectations and key management.  
> Proposals for fixing it have included deprecation of proxy 
> retargeting  
> and the use of a new SIP provisional response to inform the calling  
> party about the retargeting operation.
> 
> Security Assertion Markup Language (SAML) is an XML-based 
> standard for  
> exchanging authentication and authorization data between security  
> domains, that is, between an identity provider (a producer of  
> assertions) and a service provider (a consumer of 
> assertions). SAML is  
> a product of the OASIS Security Services Technical Committee. 
> The SIP  
> working group worked for several years on applying SAML to SIP with  
> mixed results, but the technique appears to have great promise.
> 
> 
> The initial deliverables of the RETIRE working group are:
> 
> 1) A standards-track revision of RFC 4474 that resolves its  
> limitations with respect to gateways, session border 
> controllers,  and  
> back-to-back user agents. This revision will also address the use of  
> Authenticated Identity Management in conjunction with SIP response  
> messages.
[JRE] One thing we have talked about in the past has been an additional
parameter in a SIP URI. I am not saying we definitely need to do this,
but I would not want to close the door on this if the group concludes it
would be a good thing. Another possible deliverable might a some sort of
BCP to do with E.164 numbers in SIP URIs.

> 
> 2) Specify a solution for the Unanticipated Respondent Problem in SIP.
> 
> 3) Guidelines for the use of SAML with SIP for the purposes of  
> expressing identity and role, specifically in the contexts of #1 and  
> #2 above.
> 
> 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


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Date: Wed, 25 Jun 2008 08:31:56 +0100
From: "Elwell, John" <john.elwell@siemens.com>
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	Dean Willis <dean.willis@softarmor.com>, sip@ietf.org
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Cc: Keith Drage <drage@alcatel-lucent.com>
Subject: Re: [Sip] Let's talk about a Dublin agenda
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Those overlaps would work for me.

John 

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of Hadriel Kaplan
> Sent: 24 June 2008 21:46
> To: Dean Willis; sip@ietf.org
> Cc: Keith Drage
> Subject: Re: [Sip] Let's talk about a Dublin agenda
> 
> 
> Oy vey.  That's less time than we had in Philly, and we got 
> barely anything done there.
> 
> Why can't some RAI groups overlap - for example mediactrl or 
> xcon at the same time as speermint or drinks, or let p2psip 
> overlap with any of those 4?
> 
> -hadriel
> 
> 
> > -----Original Message-----
> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of Dean
> > Willis
> > Sent: Tuesday, June 24, 2008 3:20 PM
> > To: sip@ietf.org
> > Cc: Keith Drage
> > Subject: [Sip] Let's talk about a Dublin agenda
> >
> >
> > I've been tipped off that we may be down to as little as 
> two hours (or
> > maybe 2.5)  of meeting time in Dublin due to additional WG and BOF
> > events that didn't make it onto the earlier published agenda (like
> > MEDIACTRL).
> >
> > What are the absolutely most important things we need to talk about?
> > What is the best possible use of our very valuable and very scarce
> > time together?
> >
> > I suspect that going by-rote through the list of open milestones is
> > not the best use of our time.
> >
> > Topics that I think we might want to talk about include but are not
> > limited to:
> >
> > Agenda bash and announcements usually use 10 minutes
> >
> > 1) Using SAML for SIP.
> > -- Milestone was Dec. 2007
> > -- Author has promised new rev before meeting
> > -- Is the WG committed to doing this work?
> > -- 10 minutes?
> >
> > 2) Test Matrix for SIP to Draft Standard
> > -- Milestone was Feb 2008
> > -- How are we going to move forward?
> > -- 10 minutes
> >
> > 3) Delivery of Request URI and Parameters to UAS Through Proxy
> > -- Milestone was Feb 2008
> > -- Conversation has lagged. Do we see a workable solution?
> > -- 30 minutes?
> >
> > 4) DTLS-SRTP
> > -- Milestone was Feb 2008
> > -- Delayed by the RFC 4474/Gateway/B2BUA
> > -- Decide: Do we hold DTLS-SRTP or publish with a caveat?
> > -- EKR may not be attending, can we resolve on-list? (no time?)
> >
> > 5) Mechanisms for UA Initiated Privacy
> > -- Milestone is July 2008
> > -- Author has promised new rev before meeting
> > -- 20 minutes?
> >
> > 6) Termination of early Dialog Prior to Final Response (199)
> > -- Milestone is November 2008
> > -- Substantial list activity
> > -- 30 minutes?
> >
> > 7) Guidelines for SIP Extension Modalities
> > -- No milestone
> > -- Needs conceptual discussion
> > -- Defer until #8 below is resolved? (no time?)
> >
> > 8) The INFO solution and Paul's compromise for historical 
> registration
> > -- No milestone
> > -- Previous WG consensus to "do something"
> > -- 40 minutes?
> >
> >
> > Am I missing something that is critically important?
> >
> > Which topics do we really need to talk about in order to 
> move forward?
> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
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From: "Jesske, R" <R.Jesske@telekom.de>
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Thank you for your answer.
I see there is some ongoing discussion and it is worth to progress the draf=
t.
I support such an mechanism it is very useful.

Thanks and Best Regards

Roland =


> -----Urspr=FCngliche Nachricht-----
> Von: Dean Willis [mailto:dean.willis@softarmor.com] =

> Gesendet: Dienstag, 24. Juni 2008 18:12
> An: Jesske, Roland
> Cc: sip@ietf.org
> Betreff: Re: [Sip] Draft: draft-holmberg-sip-keep-00.txt
> =

> =

> On Jun 24, 2008, at 10:07 AM, Jesske, R wrote:
> =

> > Dear all,
> > what is the status of this draft?
> =

> We are trying to determine whether there is a constituency for the  =

> draft that justifies further WG effort in advancing it.
> =

> > Will there be some consideration of this proposal of =

> defining a keep  =

> > parameter?
> =

> Yes, that's one of the conversations we're having now.
> =

> > will this possibly incorporated into draft-ietf-sip-outbound?
> >
> =

> No. It's more or less been split off from "outbound".
> =

> Were I to do this all over again, I would have separated out the  =

> keepalive function from outbound early on. But we didn't, and =

> outbound  =

> is pretty much done.
> =

> --
> Dean
> =

> =

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Date: Wed, 25 Jun 2008 08:51:59 +0100
From: "Elwell, John" <john.elwell@siemens.com>
In-reply-to: <076c01c8d643$85c210a0$c2f0200a@cisco.com>
To: Dan Wing <dwing@cisco.com>, "Vijay K. Gurbani" <vkg@alcatel-lucent.com>
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Subject: Re: [Sip] Toward the Evolution of SIP and Related Working Groups
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Dan, 

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of Dan Wing
> Sent: 24 June 2008 22:45
> To: 'Vijay K. Gurbani'
> Cc: sip@ietf.org
> Subject: Re: [Sip] Toward the Evolution of SIP and Related 
> Working Groups
> 
> > -----Original Message-----
> > From: Vijay K. Gurbani [mailto:vkg@alcatel-lucent.com] 
> > Sent: Tuesday, June 24, 2008 2:30 PM
> > To: Dan Wing
> > Cc: sip@ietf.org
> > Subject: Re: [Sip] Toward the Evolution of SIP and Related 
> > Working Groups
> > 
> > Dan Wing wrote:
> > > That wasn't my intent.
> > 
> > Dan: OK; sorry.
> 
> No worries.
> 
> > >> Otherwise, for certificate-based authentication
> > >> between proxies, some of what you write above is discussed in
> > >> the sip-domain-certs draft.
> > > 
> > > Yes, some of it is there.  Can that draft be extended to talk 
> > > about validating the From of requests (and maybe responses, I'm
> > > not sure) that come from over a TLS-authenticated connection?  Or
> > > would that be out of scope for it?
> > 
> > I believe that validating the From would be out of scope as
> > the draft currently stands.  The draft does discuss how a
> > receiving proxy (i.e., the one that did the passive accept)
> > may authenticate the sender (see S7.4,
> > http://tools.ietf.org/html/draft-ietf-sip-domain-certs-00#sect
> > ion-7.4),
> > but this discussion does not normatively involve the From
> > header.
> 
> Right.
> 
> > Furthermore, trying to match sender identity as carried in
> > From to the TLS-level connection identity as carried in the
> > certificate does not work well in the the hop-by-hop model.
> > For instance, the sender's From may as well be user@a.com, but
> > the last upstream proxy that opened up a connection to the
> > recipient could be some intermediary (proxy.com) not in the
> > a.com domain, thus making it impossible to enforce that the
> > From must match the identity of the last upstream proxy in the
> > certificate.
> 
> Agreed.
> 
> > Did you have some alternate or new thoughts on this to see if
> > we can hash out something that would still be in the context
> > of the domain-certs draft?  As it stands, Scott and I are about
> > to submit a final post WGLC revision on it by next week.
> 
> I don't think extending domain-cert's scope would be wise --
> rather, I do think this is a separate problem.  The separate
> problem is that with *some domains* you connect to (and do
> domain-certs validation with), you trust them to claim and
> use any old "From" they want -- because with those domains
> you trust them to have done some validation of their previous
> hop, who validated their previous hop, etc.  This is how
> you would configure a SIP trunk with a service provider, for
> example.
> 
> However, if you are communicating directly with another 
> organization then you would not want to allow them to assert
> any identity they wished, because the only identity you expect 
> them to send you requests with a From that had their own 
> identity (@microsoft.com, @boeing.com, etc.) -- that is, 
> the identity of their own employees.  You do not expect them 
> to assert the identity belonging to some other company.  You 
> would not extend the transitive trust to them.
[JRE] It depends. I (john@siemens.com) receive a call from
dan@cisco.com, and because I am working at home, the siemens.com proxy
forwards it to my home via a service provider. The service provider now
needs to accept dan@cisco.com (rather than xxx@siemens.com) as the
source of the call. At home I would expect to receive dan@cisco.com as
caller ID.

John
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From: "Fischer, Kai" <kai.fischer@siemens.com>
To: "Dean Willis" <dean.willis@softarmor.com>, <sip@ietf.org>,
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Subject: Re: [Sip] A proposal for breaking the DTLS-SRTP vs RFC4474
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If it is the goal to proceed with the DTLS-SRTP framework timely and to
reach the milestone, that's the only reasonable approach. However, I
hope there will be support to fix RFC 4474 and we can address the
backwards compatibility issues.

Kai


> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of Dean Willis
> Sent: Dienstag, 24. Juni 2008 19:22
> To: sip@ietf.org; Eric Rescorla; Jason Fischl
> Cc: Cullen Jennings; Keith Drage
> Subject: [Sip] A proposal for breaking the DTLS-SRTP vs 
> RFC4474 gatewaydeadlock
> 
> 
> We've gotten stuck on a fine point in DTLS-SRTP.
> 
> The current draft-ietf-sip-dtls-srtp-framework-01 uses an RFC 4474  
> Identity header to preserve the integrity of the media key's  
> fingerprint, thereby detecting a certain class of MITM attack.
> 
> However, RFC 4474 Identity headers are of questionable validity when  
> used with protocol gateways or B2BUAs.  More or less, they're 
> capable  
> of asserting the identity of the gateway, not the identity of the  
> calling party. But the recipient has no real way to figure out which  
> is which.
> 
> We've debated at some length, and with no good result, about whether  
> we should try and fix RFC 4474. We've had some suggestions that may  
> work for B2BUAs, and some other suggestions that may work for  
> gateways, but we certainly don't have a consensus.
> 
> That leaves our chartered deliverable of DTLS-SRTP hanging, and the  
> milestone has gone past months ago.
> 
> Here's a proposal:
> 
> We add a caveat about the limitation of RFC 4474 to draft-ietf-sip- 
> dtls-srtp-framework and go ahead and advance that specification. If  
> somebody later decides to fix RFC 4474, they can do so, and if  
> necessary update DTLS-SRTP if needed.
> 
> 
> Does that work for everybody?
> 
> If we agree to it, I suggest that we move the date for WGLC of draft- 
> ietf-sip-dtls-srtp-framework to July 2008, and move the 
> milestone for  
> delivery of that doc to the IESG into September.
> 
> --
> Dean
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
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	RFC4474gatewaydeadlock
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From: "Tschofenig, Hannes (NSN - FI/Espoo)" <hannes.tschofenig@nsn.com>
To: "ext Fischer, Kai" <kai.fischer@siemens.com>,
	"Dean Willis" <dean.willis@softarmor.com>, <sip@ietf.org>,
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Subject: Re: [Sip] A proposal for breaking the DTLS-SRTP vs
	RFC4474gatewaydeadlock
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Dean, 

I like your approach. 

Ciao
Hannes
 

>-----Original Message-----
>From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
>Behalf Of ext Fischer, Kai
>Sent: 25 June, 2008 11:06
>To: Dean Willis; sip@ietf.org; Eric Rescorla; Jason Fischl
>Cc: Cullen Jennings; Keith Drage
>Subject: Re: [Sip] A proposal for breaking the DTLS-SRTP vs 
>RFC4474gatewaydeadlock
>
>If it is the goal to proceed with the DTLS-SRTP framework 
>timely and to reach the milestone, that's the only reasonable 
>approach. However, I hope there will be support to fix RFC 
>4474 and we can address the backwards compatibility issues.
>
>Kai
>
>
>> -----Original Message-----
>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
>Behalf Of 
>> Dean Willis
>> Sent: Dienstag, 24. Juni 2008 19:22
>> To: sip@ietf.org; Eric Rescorla; Jason Fischl
>> Cc: Cullen Jennings; Keith Drage
>> Subject: [Sip] A proposal for breaking the DTLS-SRTP vs
>> RFC4474 gatewaydeadlock
>> 
>> 
>> We've gotten stuck on a fine point in DTLS-SRTP.
>> 
>> The current draft-ietf-sip-dtls-srtp-framework-01 uses an RFC 4474 
>> Identity header to preserve the integrity of the media key's 
>> fingerprint, thereby detecting a certain class of MITM attack.
>> 
>> However, RFC 4474 Identity headers are of questionable validity when 
>> used with protocol gateways or B2BUAs.  More or less, 
>they're capable 
>> of asserting the identity of the gateway, not the identity of the 
>> calling party. But the recipient has no real way to figure out which 
>> is which.
>> 
>> We've debated at some length, and with no good result, about whether 
>> we should try and fix RFC 4474. We've had some suggestions that may 
>> work for B2BUAs, and some other suggestions that may work for 
>> gateways, but we certainly don't have a consensus.
>> 
>> That leaves our chartered deliverable of DTLS-SRTP hanging, and the 
>> milestone has gone past months ago.
>> 
>> Here's a proposal:
>> 
>> We add a caveat about the limitation of RFC 4474 to draft-ietf-sip- 
>> dtls-srtp-framework and go ahead and advance that specification. If 
>> somebody later decides to fix RFC 4474, they can do so, and if 
>> necessary update DTLS-SRTP if needed.
>> 
>> 
>> Does that work for everybody?
>> 
>> If we agree to it, I suggest that we move the date for WGLC 
>of draft- 
>> ietf-sip-dtls-srtp-framework to July 2008, and move the 
>milestone for 
>> delivery of that doc to the IESG into September.
>> 
>> --
>> Dean
>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use 
>> sip-implementors@cs.columbia.edu for questions on current sip Use 
>> sipping@ietf.org for new developments on the application of sip
>> 
>_______________________________________________
>Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol Use 
>sip-implementors@cs.columbia.edu for questions on current sip 
>Use sipping@ietf.org for new developments on the application of sip
>
_______________________________________________
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From sip-bounces@ietf.org  Wed Jun 25 01:34:45 2008
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Date: Wed, 25 Jun 2008 09:34:42 +0100
From: "Elwell, John" <john.elwell@siemens.com>
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Subject: Re: [Sip] A proposal for breaking the DTLS-SRTP vs RFC4474
	gateway	deadlock
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Dean,

Yes, that would work for me, given an intent to push ahead with trying
to fix the RFC 4474 issues, e.g., through your proposed RETIRE WG.

John


> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of Dean Willis
> Sent: 24 June 2008 18:22
> To: sip@ietf.org; Eric Rescorla; Jason Fischl
> Cc: Cullen Jennings; Keith Drage
> Subject: [Sip] A proposal for breaking the DTLS-SRTP vs 
> RFC4474 gateway deadlock
> 
> 
> We've gotten stuck on a fine point in DTLS-SRTP.
> 
> The current draft-ietf-sip-dtls-srtp-framework-01 uses an RFC 4474  
> Identity header to preserve the integrity of the media key's  
> fingerprint, thereby detecting a certain class of MITM attack.
> 
> However, RFC 4474 Identity headers are of questionable validity when  
> used with protocol gateways or B2BUAs.  More or less, they're 
> capable  
> of asserting the identity of the gateway, not the identity of the  
> calling party. But the recipient has no real way to figure out which  
> is which.
> 
> We've debated at some length, and with no good result, about whether  
> we should try and fix RFC 4474. We've had some suggestions that may  
> work for B2BUAs, and some other suggestions that may work for  
> gateways, but we certainly don't have a consensus.
> 
> That leaves our chartered deliverable of DTLS-SRTP hanging, and the  
> milestone has gone past months ago.
> 
> Here's a proposal:
> 
> We add a caveat about the limitation of RFC 4474 to draft-ietf-sip- 
> dtls-srtp-framework and go ahead and advance that specification. If  
> somebody later decides to fix RFC 4474, they can do so, and if  
> necessary update DTLS-SRTP if needed.
> 
> 
> Does that work for everybody?
> 
> If we agree to it, I suggest that we move the date for WGLC of draft- 
> ietf-sip-dtls-srtp-framework to July 2008, and move the 
> milestone for  
> delivery of that doc to the IESG into September.
> 
> --
> Dean
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


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Hi,
I see this also as useful to have the proxy-proxy behaviour covered.

BR

Roland =


> -----Urspr=FCngliche Nachricht-----
> Von: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] Im =

> Auftrag von Christer Holmberg
> Gesendet: Mittwoch, 25. Juni 2008 00:05
> An: Hadriel Kaplan; Elwell, John; DRAGE, Keith (Keith); sip@ietf.org
> Betreff: Re: [Sip] Progress draft-holmberg-sip-keep
> =

> =

> Hi,
> =

> As I said before, I would be happy to extend the scope to =

> also cover proxy-proxy. I have also received an off-line =

> comment indicating that it would be useful. I don't think it =

> would change the mechanism itself, so...
> =

> So, if we choose to move forward with the draft I would then =

> update the scope etc it in the next version.
> =

> Does that sound ok?
> =

> Regards,
> =

> Christer =

> =

> -----Original Message-----
> From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com] =

> Sent: 24. kes=E4kuuta 2008 19:11
> To: Elwell, John; DRAGE, Keith (Keith); sip@ietf.org
> Cc: Christer Holmberg
> Subject: RE: [Sip] Progress draft-holmberg-sip-keep
> =

> =

> =

> Those cases are the ones I said would be useful in my =

> original email far below.  I.e., when I said "two reasons not =

> cited: PBX connections a la SIP trunks, and proxy-proxy =

> connections."  They are not identical, in the sense that an =

> IP-PBX can be a b2bua/UA/whatever, but also in the sense of =

> their example use-case, for example an IP-PBX could be behind =

> a NAT and thus want this keep-alive for more reasons than a =

> proxy-proxy.
> =

> -hadriel
> =

> > -----Original Message-----
> > From: Elwell, John [mailto:john.elwell@siemens.com]
> > Sent: Tuesday, June 24, 2008 11:48 AM
> > To: Hadriel Kaplan; DRAGE, Keith (Keith); sip@ietf.org
> > Cc: Christer Holmberg
> > Subject: RE: [Sip] Progress draft-holmberg-sip-keep
> >
> > Hadriel,
> >
> > That trunking case is no different from other trunking cases, e.g., =

> > proxy-to-proxy, and I have already suggested to Christer =

> that he might =

> > extend the scope of the draft to cover that. It would be =

> good to have =

> > some discussion on whether keep-alive on "trunking" =

> interfaces would =

> > be beneficial - I have an open mind on that at present.
> >
> > John
> >
> > > -----Original Message-----
> > > From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com]
> > > Sent: 24 June 2008 16:32
> > > To: Elwell, John; DRAGE, Keith (Keith); sip@ietf.org
> > > Cc: Christer Holmberg
> > > Subject: RE: [Sip] Progress draft-holmberg-sip-keep
> > >
> > > Hey John,
> > > For the case where the IP-PBX registers into the SP, =

> outbound can be =

> > > used.  But it's the other case I was talking about:
> > > where the IP-PBX does NOT register into the SP, but it =

> instead is a =

> > > trunk.
> > >
> > > -hadriel
> > >
> > >
> > > > -----Original Message-----
> > > > From: Elwell, John [mailto:john.elwell@siemens.com]
> > > > Sent: Tuesday, June 24, 2008 3:15 AM
> > > > To: Hadriel Kaplan; DRAGE, Keith (Keith); sip@ietf.org
> > > > Cc: Christer Holmberg
> > > > Subject: RE: [Sip] Progress draft-holmberg-sip-keep
> > > >
> > > > Hadriel,
> > > >
> > > > Concerning your "PBX connections a la SIP trunks" use =

> case, I am =

> > > > not convinced of this. ETSI TISPAN has specified two ways for
> > > an IPPBX to
> > > > connect to a service provider. One is the so-called
> > > subscription-based
> > > > approach, where the IPPBX registers with the SP and
> > > communicates via an
> > > > edge proxy. In this case, why not use SIP-outbound? The =

> other is =

> > > > the so-called peering-based approach, which is essentially the
> > > same as any
> > > > SIP "trunk", e.g., proxy-to-proxy, B2BUA-to-B2BUA, =

> proxy-to-gateway.
> > > > SIP-outbound does not apply to these situations, and =

> similarly the =

> > > > keep-alive mechanism is not specified for this cases. The
> > > requirements
> > > > in SIP-keep do not cover these situations.
> > > >
> > > > Basically I am not sold on the idea of a separate =

> SIP-keep spec - =

> > > > I don't think it would be the best use of WG time.
> > > >
> > > > John
> > > >
> > > >
> > > > > -----Original Message-----
> > > > > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On =

> > > > > Behalf Of Hadriel Kaplan
> > > > > Sent: 20 June 2008 18:20
> > > > > To: DRAGE, Keith (Keith); sip@ietf.org
> > > > > Cc: Christer Holmberg
> > > > > Subject: Re: [Sip] Progress draft-holmberg-sip-keep
> > > > >
> > > > >
> > > > > My 2 cents: it is a useful draft.  Personally, I =

> would like to =

> > > > > have it available for two reasons not cited: PBX =

> connections a =

> > > > > la SIP trunks, and proxy-proxy connections.
> > > > > Today I think a lot of people are using OPTIONS requests or =

> > > > > proprietary means to perform such keep-alives when =

> registration =

> > > > > is not appropriate, but it has led to some interop issues and =

> > > > > performance concerns in some cases.
> > > > >
> > > > > Since the contentious issue of what form the keep-alives take =

> > > > > have already been agreed on for outbound, this seems like a =

> > > > > simple draft to get done. (famous last words, I know)
> > > > >
> > > > > -hadriel
> > > > >
> > > > >
> > > > > > -----Original Message-----
> > > > > > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
> > > > > Behalf Of
> > > > > > DRAGE, Keith (Keith)
> > > > > > Sent: Thursday, June 19, 2008 5:31 AM
> > > > > > To: sip@ietf.org
> > > > > > Cc: Christer Holmberg
> > > > > > Subject: [Sip] Progress draft-holmberg-sip-keep
> > > > > >
> > > > > > (As SIP WG cochair)
> > > > > >
> > > > > > We have been asked by the author of
> > > > > >
> > > > > >
> > > http://www.ietf.org/internet-drafts/draft-holmberg-sip-keep-01.txt
> > > > > >
> > > > > > Whether the SIP WG can progress this document.
> > > > > >
> > > > > > Because this draft arose as a result of the discussion of
> > > > > outbound, and
> > > > > > indeed seems to reuse the requirements from outbound, and =

> > > > > > these requirements never really got handled in the SIPPING
> > > WG, it has been
> > > > > > agreed with the SIPPING chairs that we will handle this
> > > > > entirely within
> > > > > > SIP.
> > > > > >
> > > > > > Now in order to ask for charter milestones, and indeed when
> > > > > we finally
> > > > > > present this to IESG, we will be asked for the level of
> > > > > support in the
> > > > > > WG, which is also predicated on does this fix a real
> > > > > problem, or is it
> > > > > > just a corner case with limited application. So:
> > > > > >
> > > > > > QUESTION 1 TO SIP WG: Are the use cases sufficiently
> > > important to
> > > > > > proceed with this draft? The document states:
> > > > > >
> > > > > >    Chapter 3.5 of draft-ietf-sip-outbound-13
> > > [I-D.ietf-sip-outbound]
> > > > > >    defines two keep-alive techniques.  Even though the
> > > keep-alive
> > > > > >    techniques are separated from the Outbound mechanism
> > > > > >    [I-D.ietf-sip-outbound], it is currently not
> > > possible to indicate
> > > > > >    support of the keep-alive techniques without also
> > > > > indicating support
> > > > > >    for the Outbound mechanism.
> > > > > >
> > > > > >    The Outbound mechanism is enabled during the UA
> > > > > registration phase.
> > > > > >    However, there are use-cases where the UA does not
> > > > > register itself,
> > > > > >    but still needs to be able to make calls and maintain
> > > > > NAT bindings
> > > > > >    open during the duration of that call.  A =

> typical example is
> > > > > >    emergency calls.  There are also cases where entities do
> > > > > not support
> > > > > >    the Outbound mechanism, but still want to be able to
> > > > > indicate support
> > > > > >    and use the keep-alive techniques defined in
> > > > > [I-D.ietf-sip-outbound].
> > > > > >
> > > > > > At first sight this is not the most inspiring declaration
> > > > > of the need
> > > > > > for the document. Please respond indicating whether you
> > > > > consider this a
> > > > > > useful draft, and propose text that you think would be
> > > > > useful in this
> > > > > > section. Conversely, if you think this draft is not useful
> > > > > and the WG
> > > > > > has other more important things to work on first, please
> > > > > also respond.
> > > > > >
> > > > > > QUESTION 2 TO SIP WG: Do we have a robust set of
> > > requirements for
> > > > > > proceeding with this work? The document currently lists:
> > > > > >
> > > > > >    REQ 1: It MUST be possible for a UA to indicate support
> > > > > of the keep-
> > > > > >    alive techniques defined [I-D.ietf-sip-outbound] if the
> > > > > UA supports
> > > > > >    only the keep-alive part of [I-D.ietf-sip-outbound].
> > > > > >
> > > > > >    REQ 2: It MUST be possible for an edge proxy to indicate
> > > > > support of
> > > > > >    the keep-alive techniques defined
> > > > > [I-D.ietf-sip-outbound] if the edge
> > > > > >    poxy supports only the keep-alive part of
> > > > > [I-D.ietf-sip-outbound].
> > > > > >
> > > > > > It would be desirable to agree these at the outset, and not
> > > > > revisit them
> > > > > > if we continue with the work. So if you require =

> clarification, =

> > > > > > modification, or addition to these two requirements, then
> > > > > please also
> > > > > > response with your questions and proposals.
> > > > > >
> > > > > > I suggest we would like responses by 30th June 2008 in
> > > > > order to allow
> > > > > > the author to revise the document before the deadlines.
> > > > > Please note that
> > > > > > we are looking to make this decision on the list within
> > > > > this deadline
> > > > > > based on responses received, not leave it until the
> > > Dublin meeting.
> > > > > >
> > > > > > Regards
> > > > > >
> > > > > > Keith
> > > > > > _______________________________________________
> > > > > > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > > > > > This list is for NEW development of the core SIP =

> Protocol Use =

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

> > > > > sip-implementors@cs.columbia.edu for questions on current sip =

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

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


From sip-bounces@ietf.org  Wed Jun 25 02:36:43 2008
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From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
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Cc: Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Sip] Progress draft-holmberg-sip-keep
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(As SIP WG cochair)

While I am seeing some traffic on the list, I am not yet seeing enough
clear answers to the two questions asked to progress this to the AD as a
charter request.

Can I ask people who have already participated in the discussion to
repost in regard to the questions asked.

Can I ask more people in the group to review and and post their opinions
on the two questions.

Additionally, one issue that has been raised is proxy to proxy usage. If
this draft progresses, is that an additional complexity that should be
in or out?

Regards

Keith

> -----Original Message-----
> From: DRAGE, Keith (Keith) 
> Sent: Wednesday, June 18, 2008 11:28 PM
> To: sip@ietf.org
> Cc: 'Christer Holmberg'
> Subject: Progress draft-holmberg-sip-keep
> 
> (As SIP WG cochair)
> 
> We have been asked by the author of 
> 
> http://www.ietf.org/internet-drafts/draft-holmberg-sip-keep-01.txt
> 
> Whether the SIP WG can progress this document.
> 
> Because this draft arose as a result of the discussion of 
> outbound, and indeed seems to reuse the requirements from 
> outbound, and these requirements never really got handled in 
> the SIPPING WG, it has been agreed with the SIPPING chairs 
> that we will handle this entirely within SIP.
> 
> Now in order to ask for charter milestones, and indeed when 
> we finally present this to IESG, we will be asked for the 
> level of support in the WG, which is also predicated on does 
> this fix a real problem, or is it just a corner case with 
> limited application. So:
> 
> QUESTION 1 TO SIP WG: Are the use cases sufficiently 
> important to proceed with this draft? The document states:
> 
>    Chapter 3.5 of draft-ietf-sip-outbound-13 [I-D.ietf-sip-outbound]
>    defines two keep-alive techniques.  Even though the keep-alive
>    techniques are separated from the Outbound mechanism
>    [I-D.ietf-sip-outbound], it is currently not possible to indicate
>    support of the keep-alive techniques without also 
> indicating support
>    for the Outbound mechanism.
> 
>    The Outbound mechanism is enabled during the UA registration phase.
>    However, there are use-cases where the UA does not register itself,
>    but still needs to be able to make calls and maintain NAT bindings
>    open during the duration of that call.  A typical example is
>    emergency calls.  There are also cases where entities do 
> not support
>    the Outbound mechanism, but still want to be able to 
> indicate support
>    and use the keep-alive techniques defined in 
> [I-D.ietf-sip-outbound].
> 
> At first sight this is not the most inspiring declaration of 
> the need for the document. Please respond indicating whether 
> you consider this a useful draft, and propose text that you 
> think would be useful in this section. Conversely, if you 
> think this draft is not useful and the WG has other more 
> important things to work on first, please also respond.
> 
> QUESTION 2 TO SIP WG: Do we have a robust set of requirements 
> for proceeding with this work? The document currently lists:
> 
>    REQ 1: It MUST be possible for a UA to indicate support of 
> the keep-
>    alive techniques defined [I-D.ietf-sip-outbound] if the UA supports
>    only the keep-alive part of [I-D.ietf-sip-outbound].
> 
>    REQ 2: It MUST be possible for an edge proxy to indicate support of
>    the keep-alive techniques defined [I-D.ietf-sip-outbound] 
> if the edge
>    poxy supports only the keep-alive part of [I-D.ietf-sip-outbound].
> 
> It would be desirable to agree these at the outset, and not 
> revisit them if we continue with the work. So if you require 
> clarification, modification, or addition to these two 
> requirements, then please also response with your questions 
> and proposals.
> 
> I suggest we would like responses by 30th June 2008 in order 
> to allow the author to revise the document before the 
> deadlines. Please note that we are looking to make this 
> decision on the list within this deadline based on responses 
> received, not leave it until the Dublin meeting.
> 
> Regards
> 
> Keith
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


From sip-bounces@ietf.org  Wed Jun 25 02:49:23 2008
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Date: Wed, 25 Jun 2008 10:48:40 +0100
From: "Elwell, John" <john.elwell@siemens.com>
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Subject: Re: [Sip] Progress draft-holmberg-sip-keep
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Keith,

In reply to question 1, I don't see the use cases as presently
documented as being sufficiently important. I would be prepared to
reconsider this if the draft were to be extended to cover interfaces
other than UA-to-edge-proxy, in particular proxy-to-proxy. I would need
to see convincing use cases for this, plus discussion as to whether the
same technical solution applies or whether there are differences (e.g.,
who is responsible for keep-alives in symmetrical situations?).

Therefore I can't answer question 2 at present.

John 

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of DRAGE, Keith (Keith)
> Sent: 25 June 2008 10:37
> To: sip@ietf.org
> Cc: Christer Holmberg
> Subject: Re: [Sip] Progress draft-holmberg-sip-keep
> 
> (As SIP WG cochair)
> 
> While I am seeing some traffic on the list, I am not yet seeing enough
> clear answers to the two questions asked to progress this to 
> the AD as a
> charter request.
> 
> Can I ask people who have already participated in the discussion to
> repost in regard to the questions asked.
> 
> Can I ask more people in the group to review and and post 
> their opinions
> on the two questions.
> 
> Additionally, one issue that has been raised is proxy to 
> proxy usage. If
> this draft progresses, is that an additional complexity that should be
> in or out?
> 
> Regards
> 
> Keith
> 
> > -----Original Message-----
> > From: DRAGE, Keith (Keith) 
> > Sent: Wednesday, June 18, 2008 11:28 PM
> > To: sip@ietf.org
> > Cc: 'Christer Holmberg'
> > Subject: Progress draft-holmberg-sip-keep
> > 
> > (As SIP WG cochair)
> > 
> > We have been asked by the author of 
> > 
> > http://www.ietf.org/internet-drafts/draft-holmberg-sip-keep-01.txt
> > 
> > Whether the SIP WG can progress this document.
> > 
> > Because this draft arose as a result of the discussion of 
> > outbound, and indeed seems to reuse the requirements from 
> > outbound, and these requirements never really got handled in 
> > the SIPPING WG, it has been agreed with the SIPPING chairs 
> > that we will handle this entirely within SIP.
> > 
> > Now in order to ask for charter milestones, and indeed when 
> > we finally present this to IESG, we will be asked for the 
> > level of support in the WG, which is also predicated on does 
> > this fix a real problem, or is it just a corner case with 
> > limited application. So:
> > 
> > QUESTION 1 TO SIP WG: Are the use cases sufficiently 
> > important to proceed with this draft? The document states:
> > 
> >    Chapter 3.5 of draft-ietf-sip-outbound-13 [I-D.ietf-sip-outbound]
> >    defines two keep-alive techniques.  Even though the keep-alive
> >    techniques are separated from the Outbound mechanism
> >    [I-D.ietf-sip-outbound], it is currently not possible to indicate
> >    support of the keep-alive techniques without also 
> > indicating support
> >    for the Outbound mechanism.
> > 
> >    The Outbound mechanism is enabled during the UA 
> registration phase.
> >    However, there are use-cases where the UA does not 
> register itself,
> >    but still needs to be able to make calls and maintain 
> NAT bindings
> >    open during the duration of that call.  A typical example is
> >    emergency calls.  There are also cases where entities do 
> > not support
> >    the Outbound mechanism, but still want to be able to 
> > indicate support
> >    and use the keep-alive techniques defined in 
> > [I-D.ietf-sip-outbound].
> > 
> > At first sight this is not the most inspiring declaration of 
> > the need for the document. Please respond indicating whether 
> > you consider this a useful draft, and propose text that you 
> > think would be useful in this section. Conversely, if you 
> > think this draft is not useful and the WG has other more 
> > important things to work on first, please also respond.
> > 
> > QUESTION 2 TO SIP WG: Do we have a robust set of requirements 
> > for proceeding with this work? The document currently lists:
> > 
> >    REQ 1: It MUST be possible for a UA to indicate support of 
> > the keep-
> >    alive techniques defined [I-D.ietf-sip-outbound] if the 
> UA supports
> >    only the keep-alive part of [I-D.ietf-sip-outbound].
> > 
> >    REQ 2: It MUST be possible for an edge proxy to indicate 
> support of
> >    the keep-alive techniques defined [I-D.ietf-sip-outbound] 
> > if the edge
> >    poxy supports only the keep-alive part of 
> [I-D.ietf-sip-outbound].
> > 
> > It would be desirable to agree these at the outset, and not 
> > revisit them if we continue with the work. So if you require 
> > clarification, modification, or addition to these two 
> > requirements, then please also response with your questions 
> > and proposals.
> > 
> > I suggest we would like responses by 30th June 2008 in order 
> > to allow the author to revise the document before the 
> > deadlines. Please note that we are looking to make this 
> > decision on the list within this deadline based on responses 
> > received, not leave it until the Dublin meeting.
> > 
> > Regards
> > 
> > Keith
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
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From: Paul Kyzivat <pkyzivat@cisco.com>
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To: "Elwell, John" <john.elwell@siemens.com>
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Cc: sip@ietf.org, Dan Wing <dwing@cisco.com>
Subject: Re: [Sip] Toward the Evolution of SIP and Related Working Groups
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Elwell, John wrote:

>> However, if you are communicating directly with another 
>> organization then you would not want to allow them to assert
>> any identity they wished, because the only identity you expect 
>> them to send you requests with a From that had their own 
>> identity (@microsoft.com, @boeing.com, etc.) -- that is, 
>> the identity of their own employees.  You do not expect them 
>> to assert the identity belonging to some other company.  You 
>> would not extend the transitive trust to them.
> [JRE] It depends. I (john@siemens.com) receive a call from
> dan@cisco.com, and because I am working at home, the siemens.com proxy
> forwards it to my home via a service provider. The service provider now
> needs to accept dan@cisco.com (rather than xxx@siemens.com) as the
> source of the call. At home I would expect to receive dan@cisco.com as
> caller ID.

John,

I know that is what you would like to see on your phone.
But that doesn't mean there is a plausible way for it to happen.

We are discussing transitive trust here. Why would the SP trust 
siemens.com when it asserts a cisco.com domain? If that turns out to be 
incorrect, and you complain, you could justifiably complain that the SP 
lied to you. They put themself at risk by trusting such an assertion 
from siemens.

	Thanks,
	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
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Date: Wed, 25 Jun 2008 13:39:52 +0100
From: "Elwell, John" <john.elwell@siemens.com>
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Paul,

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
> Sent: 25 June 2008 13:05
> To: Elwell, John
> Cc: Dan Wing; Vijay K. Gurbani; sip@ietf.org
> Subject: Re: [Sip] Toward the Evolution of SIP and Related 
> Working Groups
> 
> 
> 
> Elwell, John wrote:
> 
> >> However, if you are communicating directly with another 
> >> organization then you would not want to allow them to assert
> >> any identity they wished, because the only identity you expect 
> >> them to send you requests with a From that had their own 
> >> identity (@microsoft.com, @boeing.com, etc.) -- that is, 
> >> the identity of their own employees.  You do not expect them 
> >> to assert the identity belonging to some other company.  You 
> >> would not extend the transitive trust to them.
> > [JRE] It depends. I (john@siemens.com) receive a call from
> > dan@cisco.com, and because I am working at home, the 
> siemens.com proxy
> > forwards it to my home via a service provider. The service 
> provider now
> > needs to accept dan@cisco.com (rather than xxx@siemens.com) as the
> > source of the call. At home I would expect to receive 
> dan@cisco.com as
> > caller ID.
> 
> John,
> 
> I know that is what you would like to see on your phone.
> But that doesn't mean there is a plausible way for it to happen.
> 
> We are discussing transitive trust here. Why would the SP trust 
> siemens.com when it asserts a cisco.com domain? If that turns 
> out to be 
> incorrect, and you complain, you could justifiably complain 
> that the SP 
> lied to you. They put themself at risk by trusting such an assertion 
> from siemens.
[JRE] I am not expecting the service provider to trust siemens.com,
because siemens.com is not making the assertion. The assertion from
cisco.com is merely being passed on by siemens.com to the service
provider. It will be the cisco.com certificate, not the siemens.com
certificate.

Of course, siemens.com also has to make an assertion, it has to assert
that it is siemens.com that is passing this request to the service
provider, and this can be done using TLS authentication. But that says
nothing about the original source of request or the caller ID. We must
not get the two layers mixed up - transport layer authentication is for
a single hop and SIP layer authentication is end-to-end (or
end-domain-to-end-domain).

John 
_______________________________________________
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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: "Tschofenig, Hannes (NSN - FI/Espoo)" <hannes.tschofenig@nsn.com>
To: "ext Elwell, John" <john.elwell@siemens.com>,
	"Paul Kyzivat" <pkyzivat@cisco.com>
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Consider the following scenario:

              +-----------+            +-----------+
              |SIP        |            |SIP        |
      +------>|Proxy      |<---------->|Proxy      |<------+
      |       |Server X   |   TLS      |Server Y   |       |
      |       +-----------+            +-----------+       |
      |                                                    |
      | TLS or                                      TLS or |
      | SIP Digest                              SIP Digest |
      |                                                    |
      |                                                    |
      v                                                    v
  +-----------+                                     +-----------+
  |SIP        |                                     |SIP        |
  |User Agent |          RTP                        |User Agent |
  |Alice      | <=================================> |Bob        |
  +-----------+                                     +-----------+
 

When there are no further proxies between X and Y then Y has the
information that Alice was authenticated by X. Proxy Y would pass that
info on to Bob. Bob trusts Y.

Obviously, the two VoIP providers may have a far more complicated
infrastructure with multiple proxies but they all belong to the same
domain and could be seen from external as just one box. 

The same would not work when there are more proxies between X and Y.
However, these guys that prefer such a deployment belong more to the
chain of trust camp rather than the end-to-end / email alike peering
camp. It is rather unlikely that you get their support for getting SIP
Identity to work in their networks.

So, do we have an indication that some folks plan to use SIP Identity
for their deployment? 

Wouldn't it be better to rely on something like SIP CERT for a better
end-to-end security mechanism (ignoring for a moment that SIP CERT also
uses SIP Identity for "simplified deployment" reasons whereby one has to
state that the usage of SIP Identity for SUBSCRIBE/NOTIFY might have
different B2BUA aspects). 

Ciao
Hannes

>-----Original Message-----
>From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
>Behalf Of ext Elwell, John
>Sent: 25 June, 2008 15:40
>To: Paul Kyzivat
>Cc: sip@ietf.org; Dan Wing
>Subject: Re: [Sip] Toward the Evolution of SIP and Related 
>Working Groups
>
>Paul,
>
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>> Sent: 25 June 2008 13:05
>> To: Elwell, John
>> Cc: Dan Wing; Vijay K. Gurbani; sip@ietf.org
>> Subject: Re: [Sip] Toward the Evolution of SIP and Related Working 
>> Groups
>> 
>> 
>> 
>> Elwell, John wrote:
>> 
>> >> However, if you are communicating directly with another 
>> >> organization then you would not want to allow them to assert any 
>> >> identity they wished, because the only identity you 
>expect them to 
>> >> send you requests with a From that had their own identity 
>> >> (@microsoft.com, @boeing.com, etc.) -- that is, the identity of 
>> >> their own employees.  You do not expect them to assert 
>the identity 
>> >> belonging to some other company.  You would not extend the 
>> >> transitive trust to them.
>> > [JRE] It depends. I (john@siemens.com) receive a call from 
>> > dan@cisco.com, and because I am working at home, the
>> siemens.com proxy
>> > forwards it to my home via a service provider. The service
>> provider now
>> > needs to accept dan@cisco.com (rather than xxx@siemens.com) as the 
>> > source of the call. At home I would expect to receive
>> dan@cisco.com as
>> > caller ID.
>> 
>> John,
>> 
>> I know that is what you would like to see on your phone.
>> But that doesn't mean there is a plausible way for it to happen.
>> 
>> We are discussing transitive trust here. Why would the SP trust 
>> siemens.com when it asserts a cisco.com domain? If that turns out to 
>> be incorrect, and you complain, you could justifiably complain that 
>> the SP lied to you. They put themself at risk by trusting such an 
>> assertion from siemens.
>[JRE] I am not expecting the service provider to trust 
>siemens.com, because siemens.com is not making the assertion. 
>The assertion from cisco.com is merely being passed on by 
>siemens.com to the service provider. It will be the cisco.com 
>certificate, not the siemens.com certificate.
>
>Of course, siemens.com also has to make an assertion, it has 
>to assert that it is siemens.com that is passing this request 
>to the service provider, and this can be done using TLS 
>authentication. But that says nothing about the original 
>source of request or the caller ID. We must not get the two 
>layers mixed up - transport layer authentication is for a 
>single hop and SIP layer authentication is end-to-end (or 
>end-domain-to-end-domain).
>
>John
>_______________________________________________
>Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol Use 
>sip-implementors@cs.columbia.edu for questions on current sip 
>Use sipping@ietf.org for new developments on the application of sip
>
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From: "Fischer, Kai" <kai.fischer@siemens.com>
To: "Dean Willis" <dean.willis@softarmor.com>, <sip@ietf.org>
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Subject: Re: [Sip] First draft, RETIRE Charter
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Hi Dean,
this looks good for me - see one comment below.
Kai

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of Dean Willis
> Sent: Mittwoch, 25. Juni 2008 01:20
> To: sip@ietf.org
> Subject: [Sip] First draft, RETIRE Charter
> 
> This is a first-pass at what I think we might do with a 
> RETIRE working  
> group
> 
> ---------------
> 
> 
> The "REal Time Identity and  Role Expression" (RETIRE) 
> Working Group's  
> purpose is to standardize the expression and assertion of 
> identity and  
> roles needed for real-time communications with the IETF's Realtime  
> Applications Infrastructure (RAI) area protocols, primarily the  
> Session Initiation Protocol (SIP) and others used in 
> conjunction with  
> SIP.
> 
> By "Identity", we mean authenticated addresses of record (as defined  
> in RFC 4474 "Enhancements for Authenticated Identity 
> Management in the  
> Session Initiation Protocol". RFC 4474 provides a means by 
> which a SIP  
> server may vouch for the "From" address of a request transiting that  
> server. It further provides a cryptographic mechanism for 
> testing the  
> source of the assertion. In the typical case, this mechanism is  
> dependent on the same Public Key Infrastructure that supports secure  
> HTTP services. Since RFC 4474 provides only authentication of the  
> source of SIP request messages and it is also important to determine  
> the identity of the sender of responses, RFC 4916 "Connected 
> Identity  
> in the Session Initiation Protocol" provides authenticated 
> identity in  
> an ongoing SIP dialog by applying RFC 4474 to requests in the 
> reverse  
> direction. While adequate for some scenarios, this mechanism is not  
> useful for SIP interactions having a single non-dialog transaction,  
> such as MESSAGE requests. RFC 4474 suffers additional limitations  
> relating to its use with gateways, session border controllers, and  
> back-to-back user agents.
The whole discussion of the limitations with RFC 4474 came up in the
context of DTLS-SRTP and end-to-end security and the limited binding of
the identity and certificate fingerprint. From my point of view, this
should be added explicitly here.

> 
> By "Role", we mean aspects of a a system user that are related  
> identity, but more dynamic in nature. For example, Alice might  
> delegate her calls to Alice. A calling party, expecting to 
> reach Alice  
> but reaching Bob, needs to know that this is a reasonable thing and  
> what Bob's role in relation to Alice is. This problem is in general  
> referred to as the "unanticipated respondent problem", and it has  
> impacts both at the level of user expectations and key management.  
> Proposals for fixing it have included deprecation of proxy 
> retargeting  
> and the use of a new SIP provisional response to inform the calling  
> party about the retargeting operation.
> 
> Security Assertion Markup Language (SAML) is an XML-based 
> standard for  
> exchanging authentication and authorization data between security  
> domains, that is, between an identity provider (a producer of  
> assertions) and a service provider (a consumer of 
> assertions). SAML is  
> a product of the OASIS Security Services Technical Committee. 
> The SIP  
> working group worked for several years on applying SAML to SIP with  
> mixed results, but the technique appears to have great promise.
> 
> 
> The initial deliverables of the RETIRE working group are:
> 
> 1) A standards-track revision of RFC 4474 that resolves its  
> limitations with respect to gateways, session border 
> controllers,  and  
> back-to-back user agents. This revision will also address the use of  
> Authenticated Identity Management in conjunction with SIP response  
> messages.
> 
> 2) Specify a solution for the Unanticipated Respondent Problem in SIP.
> 
> 3) Guidelines for the use of SAML with SIP for the purposes of  
> expressing identity and role, specifically in the contexts of #1 and  
> #2 above.
> 
> 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From: "DOLLY, MARTIN C, ATTLABS" <mdolly@att.com>
To: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>, <sip@ietf.org>
Cc: Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Sip] Progress draft-holmberg-sip-keep
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Keith,

We would find this very useful. Being the mechanism can be used between
any two SIP entities, I agree with John to expand the scope to reflect
that.

We support proxy to proxy being in, as it is needed for the network
deployments.

Martin 

-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
DRAGE, Keith (Keith)
Sent: Wednesday, June 25, 2008 5:37 AM
To: sip@ietf.org
Cc: Christer Holmberg
Subject: Re: [Sip] Progress draft-holmberg-sip-keep

(As SIP WG cochair)

While I am seeing some traffic on the list, I am not yet seeing enough
clear answers to the two questions asked to progress this to the AD as a
charter request.

Can I ask people who have already participated in the discussion to
repost in regard to the questions asked.

Can I ask more people in the group to review and and post their opinions
on the two questions.

Additionally, one issue that has been raised is proxy to proxy usage. If
this draft progresses, is that an additional complexity that should be
in or out?

Regards

Keith

> -----Original Message-----
> From: DRAGE, Keith (Keith) 
> Sent: Wednesday, June 18, 2008 11:28 PM
> To: sip@ietf.org
> Cc: 'Christer Holmberg'
> Subject: Progress draft-holmberg-sip-keep
> 
> (As SIP WG cochair)
> 
> We have been asked by the author of 
> 
> http://www.ietf.org/internet-drafts/draft-holmberg-sip-keep-01.txt
> 
> Whether the SIP WG can progress this document.
> 
> Because this draft arose as a result of the discussion of 
> outbound, and indeed seems to reuse the requirements from 
> outbound, and these requirements never really got handled in 
> the SIPPING WG, it has been agreed with the SIPPING chairs 
> that we will handle this entirely within SIP.
> 
> Now in order to ask for charter milestones, and indeed when 
> we finally present this to IESG, we will be asked for the 
> level of support in the WG, which is also predicated on does 
> this fix a real problem, or is it just a corner case with 
> limited application. So:
> 
> QUESTION 1 TO SIP WG: Are the use cases sufficiently 
> important to proceed with this draft? The document states:
> 
>    Chapter 3.5 of draft-ietf-sip-outbound-13 [I-D.ietf-sip-outbound]
>    defines two keep-alive techniques.  Even though the keep-alive
>    techniques are separated from the Outbound mechanism
>    [I-D.ietf-sip-outbound], it is currently not possible to indicate
>    support of the keep-alive techniques without also 
> indicating support
>    for the Outbound mechanism.
> 
>    The Outbound mechanism is enabled during the UA registration phase.
>    However, there are use-cases where the UA does not register itself,
>    but still needs to be able to make calls and maintain NAT bindings
>    open during the duration of that call.  A typical example is
>    emergency calls.  There are also cases where entities do 
> not support
>    the Outbound mechanism, but still want to be able to 
> indicate support
>    and use the keep-alive techniques defined in 
> [I-D.ietf-sip-outbound].
> 
> At first sight this is not the most inspiring declaration of 
> the need for the document. Please respond indicating whether 
> you consider this a useful draft, and propose text that you 
> think would be useful in this section. Conversely, if you 
> think this draft is not useful and the WG has other more 
> important things to work on first, please also respond.
> 
> QUESTION 2 TO SIP WG: Do we have a robust set of requirements 
> for proceeding with this work? The document currently lists:
> 
>    REQ 1: It MUST be possible for a UA to indicate support of 
> the keep-
>    alive techniques defined [I-D.ietf-sip-outbound] if the UA supports
>    only the keep-alive part of [I-D.ietf-sip-outbound].
> 
>    REQ 2: It MUST be possible for an edge proxy to indicate support of
>    the keep-alive techniques defined [I-D.ietf-sip-outbound] 
> if the edge
>    poxy supports only the keep-alive part of [I-D.ietf-sip-outbound].
> 
> It would be desirable to agree these at the outset, and not 
> revisit them if we continue with the work. So if you require 
> clarification, modification, or addition to these two 
> requirements, then please also response with your questions 
> and proposals.
> 
> I suggest we would like responses by 30th June 2008 in order 
> to allow the author to revise the document before the 
> deadlines. Please note that we are looking to make this 
> decision on the list within this deadline based on responses 
> received, not leave it until the Dublin meeting.
> 
> Regards
> 
> Keith
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>, "sip@ietf.org"
	<sip@ietf.org>
Date: Wed, 25 Jun 2008 10:43:41 -0400
Thread-Topic: Progress draft-holmberg-sip-keep
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Q1: yes, with some minor edits to the draft to include the proxy-proxy type thing, it's clearly valuable work.

Q2: yes, I think those are the set of requirements - though I don't know why we can' expand them in the future if we decide a couple more are required.

-hadriel


> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> DRAGE, Keith (Keith)
> Sent: Wednesday, June 25, 2008 5:37 AM
> To: sip@ietf.org
> Cc: Christer Holmberg
> Subject: Re: [Sip] Progress draft-holmberg-sip-keep
>
> (As SIP WG cochair)
>
> While I am seeing some traffic on the list, I am not yet seeing enough
> clear answers to the two questions asked to progress this to the AD as a
> charter request.
>
> Can I ask people who have already participated in the discussion to
> repost in regard to the questions asked.
>
> Can I ask more people in the group to review and and post their opinions
> on the two questions.
>
> Additionally, one issue that has been raised is proxy to proxy usage. If
> this draft progresses, is that an additional complexity that should be
> in or out?
>
> Regards
>
> Keith
>
> > -----Original Message-----
> > From: DRAGE, Keith (Keith)
> > Sent: Wednesday, June 18, 2008 11:28 PM
> > To: sip@ietf.org
> > Cc: 'Christer Holmberg'
> > Subject: Progress draft-holmberg-sip-keep
> >
> > (As SIP WG cochair)
> >
> > We have been asked by the author of
> >
> > http://www.ietf.org/internet-drafts/draft-holmberg-sip-keep-01.txt
> >
> > Whether the SIP WG can progress this document.
> >
> > Because this draft arose as a result of the discussion of
> > outbound, and indeed seems to reuse the requirements from
> > outbound, and these requirements never really got handled in
> > the SIPPING WG, it has been agreed with the SIPPING chairs
> > that we will handle this entirely within SIP.
> >
> > Now in order to ask for charter milestones, and indeed when
> > we finally present this to IESG, we will be asked for the
> > level of support in the WG, which is also predicated on does
> > this fix a real problem, or is it just a corner case with
> > limited application. So:
> >
> > QUESTION 1 TO SIP WG: Are the use cases sufficiently
> > important to proceed with this draft? The document states:
> >
> >    Chapter 3.5 of draft-ietf-sip-outbound-13 [I-D.ietf-sip-outbound]
> >    defines two keep-alive techniques.  Even though the keep-alive
> >    techniques are separated from the Outbound mechanism
> >    [I-D.ietf-sip-outbound], it is currently not possible to indicate
> >    support of the keep-alive techniques without also
> > indicating support
> >    for the Outbound mechanism.
> >
> >    The Outbound mechanism is enabled during the UA registration phase.
> >    However, there are use-cases where the UA does not register itself,
> >    but still needs to be able to make calls and maintain NAT bindings
> >    open during the duration of that call.  A typical example is
> >    emergency calls.  There are also cases where entities do
> > not support
> >    the Outbound mechanism, but still want to be able to
> > indicate support
> >    and use the keep-alive techniques defined in
> > [I-D.ietf-sip-outbound].
> >
> > At first sight this is not the most inspiring declaration of
> > the need for the document. Please respond indicating whether
> > you consider this a useful draft, and propose text that you
> > think would be useful in this section. Conversely, if you
> > think this draft is not useful and the WG has other more
> > important things to work on first, please also respond.
> >
> > QUESTION 2 TO SIP WG: Do we have a robust set of requirements
> > for proceeding with this work? The document currently lists:
> >
> >    REQ 1: It MUST be possible for a UA to indicate support of
> > the keep-
> >    alive techniques defined [I-D.ietf-sip-outbound] if the UA supports
> >    only the keep-alive part of [I-D.ietf-sip-outbound].
> >
> >    REQ 2: It MUST be possible for an edge proxy to indicate support of
> >    the keep-alive techniques defined [I-D.ietf-sip-outbound]
> > if the edge
> >    poxy supports only the keep-alive part of [I-D.ietf-sip-outbound].
> >
> > It would be desirable to agree these at the outset, and not
> > revisit them if we continue with the work. So if you require
> > clarification, modification, or addition to these two
> > requirements, then please also response with your questions
> > and proposals.
> >
> > I suggest we would like responses by 30th June 2008 in order
> > to allow the author to revise the document before the
> > deadlines. Please note that we are looking to make this
> > decision on the list within this deadline based on responses
> > received, not leave it until the Dublin meeting.
> >
> > Regards
> >
> > Keith
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
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From: "Jesske, R" <R.Jesske@telekom.de>
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Cc: christer.holmberg@ericsson.com
Subject: Re: [Sip] Progress draft-holmberg-sip-keep
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Ok
Even with the questions from Keith I didn't changed my mind and support aga=
in. =


For me are the requirements sufficient enough and could be if really needed=
 also extended later.
And we support the proxy-proxy case.

Some more support needed from my side?

BR Roland

> -----Urspr=FCngliche Nachricht-----
> Von: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] Im =

> Auftrag von Hadriel Kaplan
> Gesendet: Mittwoch, 25. Juni 2008 16:44
> An: DRAGE, Keith (Keith); sip@ietf.org
> Cc: Christer Holmberg
> Betreff: Re: [Sip] Progress draft-holmberg-sip-keep
> =

> =

> Q1: yes, with some minor edits to the draft to include the =

> proxy-proxy type thing, it's clearly valuable work.
> =

> Q2: yes, I think those are the set of requirements - though I =

> don't know why we can' expand them in the future if we decide =

> a couple more are required.
> =

> -hadriel
> =

> =

> > -----Original Message-----
> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On =

> Behalf Of
> > DRAGE, Keith (Keith)
> > Sent: Wednesday, June 25, 2008 5:37 AM
> > To: sip@ietf.org
> > Cc: Christer Holmberg
> > Subject: Re: [Sip] Progress draft-holmberg-sip-keep
> >
> > (As SIP WG cochair)
> >
> > While I am seeing some traffic on the list, I am not yet =

> seeing enough
> > clear answers to the two questions asked to progress this =

> to the AD as a
> > charter request.
> >
> > Can I ask people who have already participated in the discussion to
> > repost in regard to the questions asked.
> >
> > Can I ask more people in the group to review and and post =

> their opinions
> > on the two questions.
> >
> > Additionally, one issue that has been raised is proxy to =

> proxy usage. If
> > this draft progresses, is that an additional complexity =

> that should be
> > in or out?
> >
> > Regards
> >
> > Keith
> >
> > > -----Original Message-----
> > > From: DRAGE, Keith (Keith)
> > > Sent: Wednesday, June 18, 2008 11:28 PM
> > > To: sip@ietf.org
> > > Cc: 'Christer Holmberg'
> > > Subject: Progress draft-holmberg-sip-keep
> > >
> > > (As SIP WG cochair)
> > >
> > > We have been asked by the author of
> > >
> > > http://www.ietf.org/internet-drafts/draft-holmberg-sip-keep-01.txt
> > >
> > > Whether the SIP WG can progress this document.
> > >
> > > Because this draft arose as a result of the discussion of
> > > outbound, and indeed seems to reuse the requirements from
> > > outbound, and these requirements never really got handled in
> > > the SIPPING WG, it has been agreed with the SIPPING chairs
> > > that we will handle this entirely within SIP.
> > >
> > > Now in order to ask for charter milestones, and indeed when
> > > we finally present this to IESG, we will be asked for the
> > > level of support in the WG, which is also predicated on does
> > > this fix a real problem, or is it just a corner case with
> > > limited application. So:
> > >
> > > QUESTION 1 TO SIP WG: Are the use cases sufficiently
> > > important to proceed with this draft? The document states:
> > >
> > >    Chapter 3.5 of draft-ietf-sip-outbound-13 =

> [I-D.ietf-sip-outbound]
> > >    defines two keep-alive techniques.  Even though the keep-alive
> > >    techniques are separated from the Outbound mechanism
> > >    [I-D.ietf-sip-outbound], it is currently not possible =

> to indicate
> > >    support of the keep-alive techniques without also
> > > indicating support
> > >    for the Outbound mechanism.
> > >
> > >    The Outbound mechanism is enabled during the UA =

> registration phase.
> > >    However, there are use-cases where the UA does not =

> register itself,
> > >    but still needs to be able to make calls and maintain =

> NAT bindings
> > >    open during the duration of that call.  A typical example is
> > >    emergency calls.  There are also cases where entities do
> > > not support
> > >    the Outbound mechanism, but still want to be able to
> > > indicate support
> > >    and use the keep-alive techniques defined in
> > > [I-D.ietf-sip-outbound].
> > >
> > > At first sight this is not the most inspiring declaration of
> > > the need for the document. Please respond indicating whether
> > > you consider this a useful draft, and propose text that you
> > > think would be useful in this section. Conversely, if you
> > > think this draft is not useful and the WG has other more
> > > important things to work on first, please also respond.
> > >
> > > QUESTION 2 TO SIP WG: Do we have a robust set of requirements
> > > for proceeding with this work? The document currently lists:
> > >
> > >    REQ 1: It MUST be possible for a UA to indicate support of
> > > the keep-
> > >    alive techniques defined [I-D.ietf-sip-outbound] if =

> the UA supports
> > >    only the keep-alive part of [I-D.ietf-sip-outbound].
> > >
> > >    REQ 2: It MUST be possible for an edge proxy to =

> indicate support of
> > >    the keep-alive techniques defined [I-D.ietf-sip-outbound]
> > > if the edge
> > >    poxy supports only the keep-alive part of =

> [I-D.ietf-sip-outbound].
> > >
> > > It would be desirable to agree these at the outset, and not
> > > revisit them if we continue with the work. So if you require
> > > clarification, modification, or addition to these two
> > > requirements, then please also response with your questions
> > > and proposals.
> > >
> > > I suggest we would like responses by 30th June 2008 in order
> > > to allow the author to revise the document before the
> > > deadlines. Please note that we are looking to make this
> > > decision on the list within this deadline based on responses
> > > received, not leave it until the Dublin meeting.
> > >
> > > Regards
> > >
> > > Keith
> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> =

_______________________________________________
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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" <christer.holmberg@ericsson.com>
To: "Paul Kyzivat" <pkyzivat@cisco.com>
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Hi, 

>>>>One option, as proposed by Paul, is to say that a UAC shall be able
to RECEIVE reliable 199 responses.
>>>Based on my comments above, I am reconsidering the advisability of
that.
>> 
>>Well, I guess it wouldn't affect the complexness in the UAC, since it
will have to be able to handle reliable 
>>provisional responses anyway (if it indicates support of it, that is).
>
>Yes, I think it does increase the complexity in the implementation of
199 handling.
>
>You are right that the PRACK itself would be handled by the standard
machinery for reliable provisionals. But then 
>tearing down the dialog, which is what the 199 is supposed to trigger,
can't be done upon receipt of a reliable 199. 
>Rather it must wait until the after the PRACK is at least send, and
perhaps until the 200 for the prack is received. And >this is different
than for the unreliable 199. This isn't rocket science, but it is more
complexity. And of course it 
>requires something that sends reliable 199s in order to test the UAC.

It is true that the PRACK transaction must be "alive" until the 200 OK
is received, but I assume all other "resources" can be released once the
199 is received.

Regards,

Christer


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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Jeroen van Bemmel" <jbemmel@zonnet.nl>,
	"Rockson Li (zhengyli)" <zhengyli@cisco.com>
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Hi Jeroen,

Forgive me if I don't understand what you are trying to say, but if you hav=
e an early dialog it means that you have received a non-100 provisional res=
ponse with a to-tag.

Are you asking that I should indicate which state the entity is in when it =
sends the 199? Do you have some wording proposal?

Regards,

Christer =


-----Original Message-----
From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl] =

Sent: 24. kes=E4kuuta 2008 21:23
To: Rockson Li (zhengyli)
Cc: Christer Holmberg; sip@ietf.org
Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00

Christer,

My point was to formalize the moment a 199 is sent by relating it to the
RFC3261 ICT state machine (BTW we're all silently assuming non-INVITE reque=
sts don't use provisional responses here) So associated with PROCEEDING-->C=
OMPLETED, but indeed only if a to-tag was received before (so not only 100)

Regards,
Jeroen

PS perhaps an idea to explicitly disallow 199 for non-INVITE requests?

Rockson Li (zhengyli) wrote:
> Yes, it's a typo, 100 does not establish early-dialog.
> -Rockson
>
> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: Tuesday, June 24, 2008 6:32 PM
> To: Rockson Li (zhengyli); Jeroen van Bemmel
> Cc: sip@ietf.org
> Subject: RE: [Sip] Draft submission: draft-ietf-sip-199-00
>
>
> Hi,
>
>   =

>>> Should the 199 response contain a Contact header? And if yes, in =

>>> case a proxy sends it, should it contain the address of that proxy (sin=
ce the UAS already sent a final response)?
>>>       =

>> IF the 199 is sent reliably be the proxy must contain a Contact header c=
ontaining the address of the proxy, yes.
>>
>>     =

>>> Should we say that a proxy may only generate and send a 199 when it =

>>> receives a final error response on an INVITE client Transaction =

>>> which was in the PROCEEDING state? (i.e. 1xx response was received =

>>> before, so conceptually sending the 199 response is an action =

>>> associated with the transition from PROCEEDING to COMPLETED)
>>>       =

>> I am not sure I understand. The idea IS to send 199 when a final =

>> error response is received by the forking proxy, if a 18x has previously=
 been received.
>> [Rockson] PROCEEDING does not mean early-dialog is established, 100 =

>> Trying also move  INVITE client Transaction to PROCEEDING state.
>>     =

>
> [Christer] 199 is only sent when an early dialog has been established. 10=
0 Trying does not establish an early dialog.
>
> Regards,
>
> Christer
>
>
>
>
>
>
> Also, a forking proxy with multiple INVITE client Transaction may receive=
/forward 180 from one of them, and receive no provisional resp and final re=
sp directly from the other one, so  INVITE client Transaction's state is no=
t dependable. The decision making must be done in TU.
>
>
>
>
>
> Christer Holmberg wrote:
>   =

>> Hi,
>>
>> I agree it may be a good idea to not forbid sending it reliably.
>>
>> I do think it would be good to have text, saying that it can be sent unr=
eliable even if reliable responses are required, though, so that proxies ar=
en't forced to terminate PRACKs etc.
>>
>> Regards,
>>
>> Christer
>>  =

>>
>> -----Original Message-----
>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of =

>> Dale.Worley@comcast.net
>> Sent: 21. kes=E4kuuta 2008 1:38
>> To: sip@ietf.org
>> Subject: Re: [Sip] Draft submission: draft-ietf-sip-199-00
>>
>>    From: "Christer Holmberg" <christer.holmberg@ericsson.com>
>>
>>    [CHH] Whether the text should be in the document at all depends on if=
 we
>>    allow 199 to be sent reliably in the first place. Based on the commen=
ts
>>    received so far we should not mandate 199 to be sent reliably, even if
>>    100rel is required by the UAC. But, the question if whether we want to
>>    FORBID sending it reliably.
>>
>> If we ever might allow 199 to be used for HERFP, we should admit the pos=
sibility of sending it reliably in the first draft.  Otherwise, we'll be lo=
cked out of sending it reliably in the future.
>>
>> Dale
>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use =

>> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use =

>> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

>>     =

> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use =

> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

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> > > Furthermore, trying to match sender identity as carried in
> > > From to the TLS-level connection identity as carried in the
> > > certificate does not work well in the the hop-by-hop model.
> > > For instance, the sender's From may as well be user@a.com, but
> > > the last upstream proxy that opened up a connection to the
> > > recipient could be some intermediary (proxy.com) not in the
> > > a.com domain, thus making it impossible to enforce that the
> > > From must match the identity of the last upstream proxy in the
> > > certificate.
> > 
> > Agreed.
> > 
> > > Did you have some alternate or new thoughts on this to see if
> > > we can hash out something that would still be in the context
> > > of the domain-certs draft?  As it stands, Scott and I are about
> > > to submit a final post WGLC revision on it by next week.
> > 
> > I don't think extending domain-cert's scope would be wise --
> > rather, I do think this is a separate problem.  The separate
> > problem is that with *some domains* you connect to (and do
> > domain-certs validation with), you trust them to claim and
> > use any old "From" they want -- because with those domains
> > you trust them to have done some validation of their previous
> > hop, who validated their previous hop, etc.  This is how
> > you would configure a SIP trunk with a service provider, for
> > example.
> > 
> > However, if you are communicating directly with another 
> > organization then you would not want to allow them to assert
> > any identity they wished, because the only identity you expect 
> > them to send you requests with a From that had their own 
> > identity (@microsoft.com, @boeing.com, etc.) -- that is, 
> > the identity of their own employees.  You do not expect them 
> > to assert the identity belonging to some other company.  You 
> > would not extend the transitive trust to them.
> [JRE] It depends. I (john@siemens.com) receive a call from
> dan@cisco.com, and because I am working at home, the siemens.com proxy
> forwards it to my home via a service provider. The service 
> provider now
> needs to accept dan@cisco.com (rather than xxx@siemens.com) as the
> source of the call. At home I would expect to receive dan@cisco.com as
> caller ID.

So the title of the document would be "Validating SIP From Considered
Harmful".  After writing that document, do you feel the community would
then see the need for designing a B2BUA-survivable identity mechanism??

-d

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On Jun 25, 2008, at 2:24 AM, Elwell, John wrote:

>>

> [JRE] It also suffers an additional limitation to do with E.164  
> numbers.
> So I would prefer to say:
> "RFC 4474 suffers additional limitations relating to its use with E. 
> 164
> numbers and with gateways, session border controllers, and back-to- 
> back
> user agents."
>

Does it really have issues with E,164 numbers, or only when they are  
used through gateways? For example, I thought we had agreed  that if  
the ENUM entry for an E.164 points to a domain including an  
authentication service, then that AS could reasonably issue Identity  
headers on that phone number.

>>
> [JRE] One thing we have talked about in the past has been an  
> additional
> parameter in a SIP URI. I am not saying we definitely need to do this,
> but I would not want to close the door on this if the group  
> concludes it
> would be a good thing. Another possible deliverable might a some  
> sort of
> BCP to do with E.164 numbers in SIP URIs.

That is indeed one of the proposals for alerting an RFC4474 consumer  
to the fact that the identity assertion is "weak" because it came from  
the PSTN.

General guidance on E.164s in URIs might be a reasonable thing to do.

--
Dean

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I'm pleased to announce a working group last call for the "Body  
Handling" draft:

http://www.ietf.org/internet-drafts/draft-ietf-sip-body-handling-02.txt

This draft is intended for the Standards Track.

The only minor issue I'm aware of is an outdated reference that should  
have no impact on the review:

== Outdated reference: A later version (-08) exists of draft-ietf- 
mmusic-file-transfer-mech-06

Please review the draft carefully and post your comments back to the  
list and author.

  I'd like to conclude this WGLC period by July 11, 2008.

Ok everybody, get busy reviewing!

--
Dean


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Subject: [Sip] Second draft, RETIRE Charter
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I've revised the proposed charter fora RETIRE working group based on  
list comments. This is still a very early concept, and the formation  
of such a working group is contingent on a great many things that are  
outside of my control.


--------


The "REal Time Identity and  Role Expression" (RETIRE) Working Group's  
purpose is to standardize the expression and assertion of identity and  
roles needed for real-time communications with the IETF's Realtime  
Applications Infrastructure (RAI) area protocols, primarily the  
Session Initiation Protocol (SIP) and others used in conjunction with  
SIP.

By "Identity", we mean authenticated addresses of record (as defined  
in RFC 4474 "Enhancements for Authenticated Identity Management in the  
Session Initiation Protocol". RFC 4474 provides a means by which a SIP  
server may vouch for the "From" address of a request transiting that  
server. It further provides a cryptographic mechanism for testing the  
source of the assertion. In the typical case, this mechanism is  
dependent on the same Public Key Infrastructure that supports secure  
HTTP services. Since RFC 4474 provides only authentication of the  
source of SIP request messages and it is also important to determine  
the identity of the sender of responses, RFC 4916 "Connected Identity  
in the Session Initiation Protocol" provides authenticated identity in  
an ongoing SIP dialog by applying RFC 4474 to requests in the reverse  
direction. While adequate for some scenarios, this mechanism is not  
useful for SIP interactions having a single non-dialog transaction,  
such as MESSAGE requests. RFC 4474 suffers additional limitations  
relating to its use with E.164 numbers, gateways, session border  
controllers, and back-to-back user agents.

By "Role", we mean aspects of a a system user that are related  
identity, but more dynamic in nature. For example, Alice might  
delegate her calls to Bob. A calling party, expecting to reach Alice  
but reaching Bob, needs to know that this is a reasonable thing and  
what Bob's role in relation to Alice is. This problem is in general  
referred to as the "unanticipated respondent problem", and it has  
impacts both at the level of user expectations and key management.  
Proposals for fixing it have included deprecation of proxy retargeting  
and the use of a new SIP provisional response to inform the calling  
party about the retargeting operation.

Security Assertion Markup Language (SAML) is an XML-based standard for  
exchanging authentication and authorization data between security  
domains, that is, between an identity provider (a producer of  
assertions) and a service provider (a consumer of assertions). SAML is  
a product of the OASIS Security Services Technical Committee. The SIP  
working group worked for several years on applying SAML to SIP with  
mixed results, but the technique appears to have great promise.


The initial deliverables of the RETIRE working group are:

1) A standards-track revision of RFC 4474 that resolves its  
limitations with respect to E.164 numbers, gateways, session border  
controllers,  and back-to-back user agents. This revision will also  
address the use of Authenticated Identity Management in conjunction  
with SIP response messages. It may also be necessary to document the  
impact of such changes, if any, on the DTLS-SRTP framework.

2) Specify a solution (or set of solutions)  for the Unanticipated  
Respondent Problem in SIP.

3) Guidelines for the use of E.164 numbers as SIP identifiers.

4) Guidelines for the use of SAML with SIP for the purposes of  
expressing identity and role, specifically in the contexts of #1, #2  
and #3 above.


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On 6/25/08 8:55 AM, "Dan Wing" <dwing@Cisco.COM> wrote:

> So the title of the document would be "Validating SIP From Considered
> Harmful".  After writing that document, do you feel the community would
> then see the need for designing a B2BUA-survivable identity mechanism??
> 
> -d

No mechanism can survive with B2BUAs and intermediaries in general.
That's why we have the e2e principle on the Internet.

Henry

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> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Henry Sinnreich
>
> On 6/25/08 8:55 AM, "Dan Wing" <dwing@Cisco.COM> wrote:
>
> > So the title of the document would be "Validating SIP From Considered
> > Harmful".  After writing that document, do you feel the community would
> > then see the need for designing a B2BUA-survivable identity mechanism??
> >
> > -d
>
> No mechanism can survive with B2BUAs and intermediaries in general.
> That's why we have the e2e principle on the Internet.
> Henry

That's odd, it sure looks like your email went through a bunch of intermediaries.  I wonder what it contained when it was originally sent.
;)

-hadriel

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> > So the title of the document would be "Validating SIP From 
> > Considered
> > Harmful".  After writing that document, do you feel the 
> > community would
> > then see the need for designing a B2BUA-survivable identity 
> > mechanism??
> > 
> > -d
> 
> No mechanism can survive with B2BUAs and intermediaries in general.

Two I-Ds have been proposed which will survive B2BUAs and SBCs:

  http://tools.ietf.org/html/draft-fischer-sip-e2e-sec-media-00
  http://tools.ietf.org/html/draft-wing-sip-identity-media-02

> That's why we have the e2e principle on the Internet.

TLS and IPsec work through transport- and network-layer intermediaries (NATs
and NAPTs), and NAPTs destroy the e2e principle.

-d

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> Two I-Ds have been proposed which will survive B2BUAs and SBCs:

You would have to show that what is valid for identity can
be extended to any type of application.

> TLS and IPsec work through transport- and network-layer intermediaries (NATs
> and NAPTs), and NAPTs destroy the e2e principle.

Same as above. Can this be generalized?
What complexity costs are involved?

This is an interesting discussion to pursue with the IAB.

Henry


On 6/25/08 10:17 AM, "Dan Wing" <dwing@Cisco.COM> wrote:

>>> So the title of the document would be "Validating SIP From
>>> Considered
>>> Harmful".  After writing that document, do you feel the
>>> community would
>>> then see the need for designing a B2BUA-survivable identity
>>> mechanism??
>>> 
>>> -d
>> 
>> No mechanism can survive with B2BUAs and intermediaries in general.
> 
> Two I-Ds have been proposed which will survive B2BUAs and SBCs:
> 
>   http://tools.ietf.org/html/draft-fischer-sip-e2e-sec-media-00
>   http://tools.ietf.org/html/draft-wing-sip-identity-media-02
> 
>> That's why we have the e2e principle on the Internet.
> 
> TLS and IPsec work through transport- and network-layer intermediaries (NATs
> and NAPTs), and NAPTs destroy the e2e principle.
> 
> -d
> 

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> -----Original Message-----
> From: Henry Sinnreich [mailto:hsinnrei@adobe.com] 
> Sent: Wednesday, June 25, 2008 11:31 AM
> To: Dan Wing; 'Elwell, John'; 'Vijay K. Gurbani'
> Cc: sip@ietf.org
> Subject: Re: [Sip] Toward the Evolution of SIP and Related 
> Working Groups
> 
> > Two I-Ds have been proposed which will survive B2BUAs and SBCs:
> 
> You would have to show that what is valid for identity can
> be extended to any type of application.

So that one identity works for email, for SIP, for RTSP, for
IPsec?  That sounds like the Holy Grail.

> > TLS and IPsec work through transport- and network-layer 
> > intermediaries (NATs
> > and NAPTs), and NAPTs destroy the e2e principle.
> 
> Same as above. Can this be generalized?
> What complexity costs are involved?
> 
> This is an interesting discussion to pursue with the IAB.

-d

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If your goal is to have a short, focused and constrained WG, I would pull SAML out of it.

Also, RETIRE is probably not the best of acronyms. :)
How about "Sip Identity for Generic Name and E.164 Deployments"? (SIGNED)
Or "Identity for Trust Assurance with Sip Topologies for Everyone"? (I-TASTE, which is the only way I can identify what I am Sipping :)

-hadriel


> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Dean
> Willis
> Sent: Wednesday, June 25, 2008 12:13 PM
> To: sip@ietf.org
> Subject: [Sip] Second draft, RETIRE Charter
>
> I've revised the proposed charter fora RETIRE working group based on
> list comments. This is still a very early concept, and the formation
> of such a working group is contingent on a great many things that are
> outside of my control.
>
>
> --------
>
>
> The "REal Time Identity and  Role Expression" (RETIRE) Working Group's
> purpose is to standardize the expression and assertion of identity and
> roles needed for real-time communications with the IETF's Realtime
> Applications Infrastructure (RAI) area protocols, primarily the
> Session Initiation Protocol (SIP) and others used in conjunction with
> SIP.
>
> By "Identity", we mean authenticated addresses of record (as defined
> in RFC 4474 "Enhancements for Authenticated Identity Management in the
> Session Initiation Protocol". RFC 4474 provides a means by which a SIP
> server may vouch for the "From" address of a request transiting that
> server. It further provides a cryptographic mechanism for testing the
> source of the assertion. In the typical case, this mechanism is
> dependent on the same Public Key Infrastructure that supports secure
> HTTP services. Since RFC 4474 provides only authentication of the
> source of SIP request messages and it is also important to determine
> the identity of the sender of responses, RFC 4916 "Connected Identity
> in the Session Initiation Protocol" provides authenticated identity in
> an ongoing SIP dialog by applying RFC 4474 to requests in the reverse
> direction. While adequate for some scenarios, this mechanism is not
> useful for SIP interactions having a single non-dialog transaction,
> such as MESSAGE requests. RFC 4474 suffers additional limitations
> relating to its use with E.164 numbers, gateways, session border
> controllers, and back-to-back user agents.
>
> By "Role", we mean aspects of a a system user that are related
> identity, but more dynamic in nature. For example, Alice might
> delegate her calls to Bob. A calling party, expecting to reach Alice
> but reaching Bob, needs to know that this is a reasonable thing and
> what Bob's role in relation to Alice is. This problem is in general
> referred to as the "unanticipated respondent problem", and it has
> impacts both at the level of user expectations and key management.
> Proposals for fixing it have included deprecation of proxy retargeting
> and the use of a new SIP provisional response to inform the calling
> party about the retargeting operation.
>
> Security Assertion Markup Language (SAML) is an XML-based standard for
> exchanging authentication and authorization data between security
> domains, that is, between an identity provider (a producer of
> assertions) and a service provider (a consumer of assertions). SAML is
> a product of the OASIS Security Services Technical Committee. The SIP
> working group worked for several years on applying SAML to SIP with
> mixed results, but the technique appears to have great promise.
>
>
> The initial deliverables of the RETIRE working group are:
>
> 1) A standards-track revision of RFC 4474 that resolves its
> limitations with respect to E.164 numbers, gateways, session border
> controllers,  and back-to-back user agents. This revision will also
> address the use of Authenticated Identity Management in conjunction
> with SIP response messages. It may also be necessary to document the
> impact of such changes, if any, on the DTLS-SRTP framework.
>
> 2) Specify a solution (or set of solutions)  for the Unanticipated
> Respondent Problem in SIP.
>
> 3) Guidelines for the use of E.164 numbers as SIP identifiers.
>
> 4) Guidelines for the use of SAML with SIP for the purposes of
> expressing identity and role, specifically in the contexts of #1, #2
> and #3 above.
>
>
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From: <Markus.Isomaki@nokia.com>
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Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use case
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Hi,

I'm a bit sceptical about the need for keep-alives between proxies. It
is of course entirely possible that an enterprise PBX is connected to
(or peering with) a service provider proxy through a NAT and/or a
firewall. However, wouldn't such a NAT or firewall be under the
administration of either the enterprise itself or its ISP (who quite
often would be the SIP service provider), and the required port
forwardings or firewall rules could be set through administration. This
means that there would not be need for keepalive traffic to implicitely
keep the mapping/pinhole open.

Or are there really deployment cases where there are SIP PBXs behind
unadministrated NATs or firewalls?

Wouldn't we then need keepalives for SMTP as well, or how has the e-mail
infrastructure managed to solve this problem? 

Markus 
 

>-----Original Message-----
>From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
>Behalf Of ext Elwell, John
>Sent: 24 June, 2008 10:15
>To: Hadriel Kaplan; DRAGE, Keith (Keith); sip@ietf.org
>Cc: Christer Holmberg
>Subject: Re: [Sip] Progress draft-holmberg-sip-keep
>
>Hadriel,
>
>Concerning your "PBX connections a la SIP trunks" use case, I 
>am not convinced of this. ETSI TISPAN has specified two ways 
>for an IPPBX to connect to a service provider. One is the 
>so-called subscription-based approach, where the IPPBX 
>registers with the SP and communicates via an edge proxy. In 
>this case, why not use SIP-outbound? The other is the 
>so-called peering-based approach, which is essentially the 
>same as any SIP "trunk", e.g., proxy-to-proxy, B2BUA-to-B2BUA, 
>proxy-to-gateway.
>SIP-outbound does not apply to these situations, and similarly 
>the keep-alive mechanism is not specified for this cases. The 
>requirements in SIP-keep do not cover these situations.
>
>Basically I am not sold on the idea of a separate SIP-keep 
>spec - I don't think it would be the best use of WG time.
>
>John
>
>
>> -----Original Message-----
>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
>Behalf Of 
>> Hadriel Kaplan
>> Sent: 20 June 2008 18:20
>> To: DRAGE, Keith (Keith); sip@ietf.org
>> Cc: Christer Holmberg
>> Subject: Re: [Sip] Progress draft-holmberg-sip-keep
>> 
>> 
>> My 2 cents: it is a useful draft.  Personally, I would like 
>to have it 
>> available for two reasons not cited: PBX connections a la 
>SIP trunks, 
>> and proxy-proxy connections.
>> Today I think a lot of people are using OPTIONS requests or 
>> proprietary means to perform such keep-alives when 
>registration is not 
>> appropriate, but it has led to some interop issues and performance 
>> concerns in some cases.
>> 
>> Since the contentious issue of what form the keep-alives take have 
>> already been agreed on for outbound, this seems like a 
>simple draft to 
>> get done. (famous last words, I know)
>> 
>> -hadriel
>> 
>> 
>> > -----Original Message-----
>> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
>> Behalf Of
>> > DRAGE, Keith (Keith)
>> > Sent: Thursday, June 19, 2008 5:31 AM
>> > To: sip@ietf.org
>> > Cc: Christer Holmberg
>> > Subject: [Sip] Progress draft-holmberg-sip-keep
>> >
>> > (As SIP WG cochair)
>> >
>> > We have been asked by the author of
>> >
>> > http://www.ietf.org/internet-drafts/draft-holmberg-sip-keep-01.txt
>> >
>> > Whether the SIP WG can progress this document.
>> >
>> > Because this draft arose as a result of the discussion of
>> outbound, and
>> > indeed seems to reuse the requirements from outbound, and these 
>> > requirements never really got handled in the SIPPING WG, 
>it has been 
>> > agreed with the SIPPING chairs that we will handle this
>> entirely within
>> > SIP.
>> >
>> > Now in order to ask for charter milestones, and indeed when
>> we finally
>> > present this to IESG, we will be asked for the level of
>> support in the
>> > WG, which is also predicated on does this fix a real
>> problem, or is it
>> > just a corner case with limited application. So:
>> >
>> > QUESTION 1 TO SIP WG: Are the use cases sufficiently important to 
>> > proceed with this draft? The document states:
>> >
>> >    Chapter 3.5 of draft-ietf-sip-outbound-13 
>[I-D.ietf-sip-outbound]
>> >    defines two keep-alive techniques.  Even though the keep-alive
>> >    techniques are separated from the Outbound mechanism
>> >    [I-D.ietf-sip-outbound], it is currently not possible 
>to indicate
>> >    support of the keep-alive techniques without also
>> indicating support
>> >    for the Outbound mechanism.
>> >
>> >    The Outbound mechanism is enabled during the UA
>> registration phase.
>> >    However, there are use-cases where the UA does not
>> register itself,
>> >    but still needs to be able to make calls and maintain
>> NAT bindings
>> >    open during the duration of that call.  A typical example is
>> >    emergency calls.  There are also cases where entities do
>> not support
>> >    the Outbound mechanism, but still want to be able to
>> indicate support
>> >    and use the keep-alive techniques defined in
>> [I-D.ietf-sip-outbound].
>> >
>> > At first sight this is not the most inspiring declaration
>> of the need
>> > for the document. Please respond indicating whether you
>> consider this a
>> > useful draft, and propose text that you think would be
>> useful in this
>> > section. Conversely, if you think this draft is not useful
>> and the WG
>> > has other more important things to work on first, please
>> also respond.
>> >
>> > QUESTION 2 TO SIP WG: Do we have a robust set of requirements for 
>> > proceeding with this work? The document currently lists:
>> >
>> >    REQ 1: It MUST be possible for a UA to indicate support
>> of the keep-
>> >    alive techniques defined [I-D.ietf-sip-outbound] if the
>> UA supports
>> >    only the keep-alive part of [I-D.ietf-sip-outbound].
>> >
>> >    REQ 2: It MUST be possible for an edge proxy to indicate
>> support of
>> >    the keep-alive techniques defined
>> [I-D.ietf-sip-outbound] if the edge
>> >    poxy supports only the keep-alive part of
>> [I-D.ietf-sip-outbound].
>> >
>> > It would be desirable to agree these at the outset, and not
>> revisit them
>> > if we continue with the work. So if you require clarification, 
>> > modification, or addition to these two requirements, then
>> please also
>> > response with your questions and proposals.
>> >
>> > I suggest we would like responses by 30th June 2008 in
>> order to allow
>> > the author to revise the document before the deadlines. 
>> Please note that
>> > we are looking to make this decision on the list within
>> this deadline
>> > based on responses received, not leave it until the Dublin meeting.
>> >
>> > Regards
>> >
>> > Keith
>> > _______________________________________________
>> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> > This list is for NEW development of the core SIP Protocol Use 
>> > sip-implementors@cs.columbia.edu for questions on current sip Use 
>> > sipping@ietf.org for new developments on the application of sip
>> _______________________________________________
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>> This list is for NEW development of the core SIP Protocol Use 
>> sip-implementors@cs.columbia.edu for questions on current sip Use 
>> sipping@ietf.org for new developments on the application of sip
>> 
>_______________________________________________
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>Use sipping@ietf.org for new developments on the application of sip
>
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "Markus.Isomaki@nokia.com" <Markus.Isomaki@nokia.com>,
	"john.elwell@siemens.com" <john.elwell@siemens.com>,
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Thread-Topic: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use case
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Yes I am of that same opinion - that any real "IP-PBX" or whatever big enough NOT to be doing Registration, and to instead do static provisioning or DNS, would be given a static hole/DMZ address in their firewall/NAT.  But some of my customers have told me otherwise. (interestingly mostly in APAC region)
There's also some concern that while a static entry is there for inbound TCP connections, the PBX creates outbound ones to the service provider which are ephemeral port sources and need to live for very long durations (though why they can't just do TCP keepalive is beyond me, but I'm no expert).

But anyway, the big issue we've seen is that we need both the PBX and the service provider box to detect failure before an active call/request attempt is made; to trigger alternate route selection without waiting for transport failure, and as a method to detect liveness again and revert.  Today that's almost exclusively done with Options requests as far as I've seen, and lots of people don't seem to like that.

-hadriel


> -----Original Message-----
> From: Markus.Isomaki@nokia.com [mailto:Markus.Isomaki@nokia.com]
> Sent: Wednesday, June 25, 2008 3:11 PM
> To: john.elwell@siemens.com; Hadriel Kaplan; drage@alcatel-lucent.com;
> sip@ietf.org
> Cc: christer.holmberg@ericsson.com
> Subject: RE: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use
> case
>
> Hi,
>
> I'm a bit sceptical about the need for keep-alives between proxies. It
> is of course entirely possible that an enterprise PBX is connected to
> (or peering with) a service provider proxy through a NAT and/or a
> firewall. However, wouldn't such a NAT or firewall be under the
> administration of either the enterprise itself or its ISP (who quite
> often would be the SIP service provider), and the required port
> forwardings or firewall rules could be set through administration. This
> means that there would not be need for keepalive traffic to implicitely
> keep the mapping/pinhole open.
>
> Or are there really deployment cases where there are SIP PBXs behind
> unadministrated NATs or firewalls?
>
> Wouldn't we then need keepalives for SMTP as well, or how has the e-mail
> infrastructure managed to solve this problem?
>
> Markus
>
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use case
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Hadriel, Markus,

Instead of standardizing keep-alives between proxies, how about we 
define a "virtual UA" on each element (similar to the one described in 
RFC3261 section 16.7 point 6) to be used to provide this functionality? 
(using existing outbound functionality, perhaps both ways)

Regards,
Jeroen

Hadriel Kaplan wrote:
> Yes I am of that same opinion - that any real "IP-PBX" or whatever big enough NOT to be doing Registration, and to instead do static provisioning or DNS, would be given a static hole/DMZ address in their firewall/NAT.  But some of my customers have told me otherwise. (interestingly mostly in APAC region)
> There's also some concern that while a static entry is there for inbound TCP connections, the PBX creates outbound ones to the service provider which are ephemeral port sources and need to live for very long durations (though why they can't just do TCP keepalive is beyond me, but I'm no expert).
>
> But anyway, the big issue we've seen is that we need both the PBX and the service provider box to detect failure before an active call/request attempt is made; to trigger alternate route selection without waiting for transport failure, and as a method to detect liveness again and revert.  Today that's almost exclusively done with Options requests as far as I've seen, and lots of people don't seem to like that.
>
> -hadriel
>
>
>   
>> -----Original Message-----
>> From: Markus.Isomaki@nokia.com [mailto:Markus.Isomaki@nokia.com]
>> Sent: Wednesday, June 25, 2008 3:11 PM
>> To: john.elwell@siemens.com; Hadriel Kaplan; drage@alcatel-lucent.com;
>> sip@ietf.org
>> Cc: christer.holmberg@ericsson.com
>> Subject: RE: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use
>> case
>>
>> Hi,
>>
>> I'm a bit sceptical about the need for keep-alives between proxies. It
>> is of course entirely possible that an enterprise PBX is connected to
>> (or peering with) a service provider proxy through a NAT and/or a
>> firewall. However, wouldn't such a NAT or firewall be under the
>> administration of either the enterprise itself or its ISP (who quite
>> often would be the SIP service provider), and the required port
>> forwardings or firewall rules could be set through administration. This
>> means that there would not be need for keepalive traffic to implicitely
>> keep the mapping/pinhole open.
>>
>> Or are there really deployment cases where there are SIP PBXs behind
>> unadministrated NATs or firewalls?
>>
>> Wouldn't we then need keepalives for SMTP as well, or how has the e-mail
>> infrastructure managed to solve this problem?
>>
>> Markus
>>
>>     
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>   
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From: "Elwell, John" <john.elwell@siemens.com>
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Subject: Re: [Sip] Toward the Evolution of SIP and Related Working Groups
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Hannes, 

> -----Original Message-----
> From: Tschofenig, Hannes (NSN - FI/Espoo) 
> [mailto:hannes.tschofenig@nsn.com] 
> Sent: 25 June 2008 14:00
> To: Elwell, John; Paul Kyzivat
> Cc: sip@ietf.org; Dan Wing
> Subject: RE: [Sip] Toward the Evolution of SIP and Related 
> Working Groups
> 
> 
> Consider the following scenario:
> 
>               +-----------+            +-----------+
>               |SIP        |            |SIP        |
>       +------>|Proxy      |<---------->|Proxy      |<------+
>       |       |Server X   |   TLS      |Server Y   |       |
>       |       +-----------+            +-----------+       |
>       |                                                    |
>       | TLS or                                      TLS or |
>       | SIP Digest                              SIP Digest |
>       |                                                    |
>       |                                                    |
>       v                                                    v
>   +-----------+                                     +-----------+
>   |SIP        |                                     |SIP        |
>   |User Agent |          RTP                        |User Agent |
>   |Alice      | <=================================> |Bob        |
>   +-----------+                                     +-----------+
>  
> 
> When there are no further proxies between X and Y then Y has the
> information that Alice was authenticated by X. Proxy Y would pass that
> info on to Bob. Bob trusts Y.
> 
> Obviously, the two VoIP providers may have a far more complicated
> infrastructure with multiple proxies but they all belong to the same
> domain and could be seen from external as just one box. 
> 
> The same would not work when there are more proxies between X and Y.
> However, these guys that prefer such a deployment belong more to the
> chain of trust camp rather than the end-to-end / email alike peering
> camp. It is rather unlikely that you get their support for getting SIP
> Identity to work in their networks.
> 
> So, do we have an indication that some folks plan to use SIP Identity
> for their deployment? 
[JRE] Well, what about enterprise via service provider to enterprise? As
an enterprise provider, I would like to be able to use SIP identity, but
a precondition is that it gets through that intermediate service
provider(s).


> 
> Wouldn't it be better to rely on something like SIP CERT for a better
> end-to-end security mechanism (ignoring for a moment that SIP 
> CERT also
> uses SIP Identity for "simplified deployment" reasons whereby 
> one has to
> state that the usage of SIP Identity for SUBSCRIBE/NOTIFY might have
> different B2BUA aspects). 
[JRE] You have lost me there. What is SIP CERT? Can you point me at a
draft?

John
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From: "Elwell, John" <john.elwell@siemens.com>
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Dan, 

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of Dan Wing
> Sent: 25 June 2008 16:55
> To: Elwell, John; 'Vijay K. Gurbani'
> Cc: sip@ietf.org
> Subject: Re: [Sip] Toward the Evolution of SIP and Related 
> Working Groups
> 
> 
> > > > Furthermore, trying to match sender identity as carried in
> > > > From to the TLS-level connection identity as carried in the
> > > > certificate does not work well in the the hop-by-hop model.
> > > > For instance, the sender's From may as well be user@a.com, but
> > > > the last upstream proxy that opened up a connection to the
> > > > recipient could be some intermediary (proxy.com) not in the
> > > > a.com domain, thus making it impossible to enforce that the
> > > > From must match the identity of the last upstream proxy in the
> > > > certificate.
> > > 
> > > Agreed.
> > > 
> > > > Did you have some alternate or new thoughts on this to see if
> > > > we can hash out something that would still be in the context
> > > > of the domain-certs draft?  As it stands, Scott and I are about
> > > > to submit a final post WGLC revision on it by next week.
> > > 
> > > I don't think extending domain-cert's scope would be wise --
> > > rather, I do think this is a separate problem.  The separate
> > > problem is that with *some domains* you connect to (and do
> > > domain-certs validation with), you trust them to claim and
> > > use any old "From" they want -- because with those domains
> > > you trust them to have done some validation of their previous
> > > hop, who validated their previous hop, etc.  This is how
> > > you would configure a SIP trunk with a service provider, for
> > > example.
> > > 
> > > However, if you are communicating directly with another 
> > > organization then you would not want to allow them to assert
> > > any identity they wished, because the only identity you expect 
> > > them to send you requests with a From that had their own 
> > > identity (@microsoft.com, @boeing.com, etc.) -- that is, 
> > > the identity of their own employees.  You do not expect them 
> > > to assert the identity belonging to some other company.  You 
> > > would not extend the transitive trust to them.
> > [JRE] It depends. I (john@siemens.com) receive a call from
> > dan@cisco.com, and because I am working at home, the 
> siemens.com proxy
> > forwards it to my home via a service provider. The service 
> > provider now
> > needs to accept dan@cisco.com (rather than xxx@siemens.com) as the
> > source of the call. At home I would expect to receive 
> dan@cisco.com as
> > caller ID.
> 
> So the title of the document would be "Validating SIP From Considered
> Harmful".  After writing that document, do you feel the 
> community would
> then see the need for designing a B2BUA-survivable identity 
> mechanism??
[JRE] It's not the best starting point - rather a long train of thought
to get to the conclusion. However, it is one of the many factors in this
complicated equation.

John
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From: "Elwell, John" <john.elwell@siemens.com>
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Dean, 

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com] 
> Sent: 25 June 2008 17:01
> To: Elwell, John
> Cc: sip@ietf.org
> Subject: Re: [Sip] First draft, RETIRE Charter
> 
> 
> On Jun 25, 2008, at 2:24 AM, Elwell, John wrote:
> 
> >>
> 
> > [JRE] It also suffers an additional limitation to do with E.164  
> > numbers.
> > So I would prefer to say:
> > "RFC 4474 suffers additional limitations relating to its 
> use with E. 
> > 164
> > numbers and with gateways, session border controllers, and back-to- 
> > back
> > user agents."
> >
> 
> Does it really have issues with E,164 numbers, or only when they are  
> used through gateways? For example, I thought we had agreed  that if  
> the ENUM entry for an E.164 points to a domain including an  
> authentication service, then that AS could reasonably issue Identity  
> headers on that phone number.
[JRE] That was suggested - I am not sure there was an agreement, and
even that would need documenting.

John
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Thread-Topic: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use case
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Hadriel Kaplan" <HKaplan@acmepacket.com>, <Markus.Isomaki@nokia.com>,
	<john.elwell@siemens.com>, <drage@alcatel-lucent.com>, <sip@ietf.org>
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Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use case
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Hi,

Also, in some of the "proxy-to-proxy" cases it is not necessarily about the=
 need for NAT keep-alive, but more about the need for a "heartbeat" functio=
n (I guess that is what Hadriel is also talking about). Today that function=
ality is sometimes implemented using SIP OPTIONS etc, which is a rather hea=
vy and clumsy mechanism for that purpose.

Regards,

Christer

 =


-----Original Message-----
From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com] =

Sent: 25. kes=E4kuuta 2008 22:34
To: Markus.Isomaki@nokia.com; john.elwell@siemens.com; drage@alcatel-lucent=
.com; sip@ietf.org
Cc: Christer Holmberg
Subject: RE: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use case


Yes I am of that same opinion - that any real "IP-PBX" or whatever big enou=
gh NOT to be doing Registration, and to instead do static provisioning or D=
NS, would be given a static hole/DMZ address in their firewall/NAT.  But so=
me of my customers have told me otherwise. (interestingly mostly in APAC re=
gion) There's also some concern that while a static entry is there for inbo=
und TCP connections, the PBX creates outbound ones to the service provider =
which are ephemeral port sources and need to live for very long durations (=
though why they can't just do TCP keepalive is beyond me, but I'm no expert=
).

But anyway, the big issue we've seen is that we need both the PBX and the s=
ervice provider box to detect failure before an active call/request attempt=
 is made; to trigger alternate route selection without waiting for transpor=
t failure, and as a method to detect liveness again and revert.  Today that=
's almost exclusively done with Options requests as far as I've seen, and l=
ots of people don't seem to like that.

-hadriel


> -----Original Message-----
> From: Markus.Isomaki@nokia.com [mailto:Markus.Isomaki@nokia.com]
> Sent: Wednesday, June 25, 2008 3:11 PM
> To: john.elwell@siemens.com; Hadriel Kaplan; drage@alcatel-lucent.com; =

> sip@ietf.org
> Cc: christer.holmberg@ericsson.com
> Subject: RE: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy =

> use case
>
> Hi,
>
> I'm a bit sceptical about the need for keep-alives between proxies. It =

> is of course entirely possible that an enterprise PBX is connected to =

> (or peering with) a service provider proxy through a NAT and/or a =

> firewall. However, wouldn't such a NAT or firewall be under the =

> administration of either the enterprise itself or its ISP (who quite =

> often would be the SIP service provider), and the required port =

> forwardings or firewall rules could be set through administration. =

> This means that there would not be need for keepalive traffic to =

> implicitely keep the mapping/pinhole open.
>
> Or are there really deployment cases where there are SIP PBXs behind =

> unadministrated NATs or firewalls?
>
> Wouldn't we then need keepalives for SMTP as well, or how has the =

> e-mail infrastructure managed to solve this problem?
>
> Markus
>
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Jeroen van Bemmel" <jbemmel@zonnet.nl>,
	"Hadriel Kaplan" <HKaplan@acmepacket.com>
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Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use case
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Hi,

So, you are proposing that each element should be a B2BUA, and both element=
s then register towards each other and use Outbound???

Regards,

Christer =


-----Original Message-----
From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl] =

Sent: 25. kes=E4kuuta 2008 22:57
To: Hadriel Kaplan
Cc: Markus.Isomaki@nokia.com; john.elwell@siemens.com; drage@alcatel-lucent=
.com; sip@ietf.org; Christer Holmberg
Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use case

Hadriel, Markus,

Instead of standardizing keep-alives between proxies, how about we define a=
 "virtual UA" on each element (similar to the one described in
RFC3261 section 16.7 point 6) to be used to provide this functionality? =

(using existing outbound functionality, perhaps both ways)

Regards,
Jeroen

Hadriel Kaplan wrote:
> Yes I am of that same opinion - that any real "IP-PBX" or whatever big =

> enough NOT to be doing Registration, and to instead do static provisionin=
g or DNS, would be given a static hole/DMZ address in their firewall/NAT.  =
But some of my customers have told me otherwise. (interestingly mostly in A=
PAC region) There's also some concern that while a static entry is there fo=
r inbound TCP connections, the PBX creates outbound ones to the service pro=
vider which are ephemeral port sources and need to live for very long durat=
ions (though why they can't just do TCP keepalive is beyond me, but I'm no =
expert).
>
> But anyway, the big issue we've seen is that we need both the PBX and the=
 service provider box to detect failure before an active call/request attem=
pt is made; to trigger alternate route selection without waiting for transp=
ort failure, and as a method to detect liveness again and revert.  Today th=
at's almost exclusively done with Options requests as far as I've seen, and=
 lots of people don't seem to like that.
>
> -hadriel
>
>
>   =

>> -----Original Message-----
>> From: Markus.Isomaki@nokia.com [mailto:Markus.Isomaki@nokia.com]
>> Sent: Wednesday, June 25, 2008 3:11 PM
>> To: john.elwell@siemens.com; Hadriel Kaplan; =

>> drage@alcatel-lucent.com; sip@ietf.org
>> Cc: christer.holmberg@ericsson.com
>> Subject: RE: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy =

>> use case
>>
>> Hi,
>>
>> I'm a bit sceptical about the need for keep-alives between proxies. =

>> It is of course entirely possible that an enterprise PBX is connected =

>> to (or peering with) a service provider proxy through a NAT and/or a =

>> firewall. However, wouldn't such a NAT or firewall be under the =

>> administration of either the enterprise itself or its ISP (who quite =

>> often would be the SIP service provider), and the required port =

>> forwardings or firewall rules could be set through administration. =

>> This means that there would not be need for keepalive traffic to =

>> implicitely keep the mapping/pinhole open.
>>
>> Or are there really deployment cases where there are SIP PBXs behind =

>> unadministrated NATs or firewalls?
>>
>> Wouldn't we then need keepalives for SMTP as well, or how has the =

>> e-mail infrastructure managed to solve this problem?
>>
>> Markus
>>
>>     =

> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use =

> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

_______________________________________________
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This list is for NEW development of the core SIP 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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No, not a B2BUA. Each element would have a virtual UA function "in =

parallel" to its regular function (e.g. being a proxy). And indeed, one =

way to implement this would be to keep either one (one way) or two =

(opposite ways) flows active between them.

Like so:

|-----------|                     |-----------|
|   Proxy     |                     |  Proxy     |
|-----------|                     |-----------|
|     UA      | <---------->|      UA     |
|-----------|                     |-----------|

Each "UA" would implement both a simple registrar and a UAC performing =

registration. It could also be setup asymmetrically, with the (smaller) =

IP-PBX doing a single registration towards the (bigger) Service provider =

network. Registration expiry would denote a loss of connectivity.

Regards,
Jeroen


Christer Holmberg wrote:
> Hi,
>
> So, you are proposing that each element should be a B2BUA, and both eleme=
nts then register towards each other and use Outbound???
>
> Regards,
>
> Christer =

>
> -----Original Message-----
> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl] =

> Sent: 25. kes=E4kuuta 2008 22:57
> To: Hadriel Kaplan
> Cc: Markus.Isomaki@nokia.com; john.elwell@siemens.com; drage@alcatel-luce=
nt.com; sip@ietf.org; Christer Holmberg
> Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use c=
ase
>
> Hadriel, Markus,
>
> Instead of standardizing keep-alives between proxies, how about we define=
 a "virtual UA" on each element (similar to the one described in
> RFC3261 section 16.7 point 6) to be used to provide this functionality? =

> (using existing outbound functionality, perhaps both ways)
>
> Regards,
> Jeroen
>
> Hadriel Kaplan wrote:
>   =

>> Yes I am of that same opinion - that any real "IP-PBX" or whatever big =

>> enough NOT to be doing Registration, and to instead do static provisioni=
ng or DNS, would be given a static hole/DMZ address in their firewall/NAT. =
 But some of my customers have told me otherwise. (interestingly mostly in =
APAC region) There's also some concern that while a static entry is there f=
or inbound TCP connections, the PBX creates outbound ones to the service pr=
ovider which are ephemeral port sources and need to live for very long dura=
tions (though why they can't just do TCP keepalive is beyond me, but I'm no=
 expert).
>>
>> But anyway, the big issue we've seen is that we need both the PBX and th=
e service provider box to detect failure before an active call/request atte=
mpt is made; to trigger alternate route selection without waiting for trans=
port failure, and as a method to detect liveness again and revert.  Today t=
hat's almost exclusively done with Options requests as far as I've seen, an=
d lots of people don't seem to like that.
>>
>> -hadriel
>>
>>
>>   =

>>     =

>>> -----Original Message-----
>>> From: Markus.Isomaki@nokia.com [mailto:Markus.Isomaki@nokia.com]
>>> Sent: Wednesday, June 25, 2008 3:11 PM
>>> To: john.elwell@siemens.com; Hadriel Kaplan; =

>>> drage@alcatel-lucent.com; sip@ietf.org
>>> Cc: christer.holmberg@ericsson.com
>>> Subject: RE: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy =

>>> use case
>>>
>>> Hi,
>>>
>>> I'm a bit sceptical about the need for keep-alives between proxies. =

>>> It is of course entirely possible that an enterprise PBX is connected =

>>> to (or peering with) a service provider proxy through a NAT and/or a =

>>> firewall. However, wouldn't such a NAT or firewall be under the =

>>> administration of either the enterprise itself or its ISP (who quite =

>>> often would be the SIP service provider), and the required port =

>>> forwardings or firewall rules could be set through administration. =

>>> This means that there would not be need for keepalive traffic to =

>>> implicitely keep the mapping/pinhole open.
>>>
>>> Or are there really deployment cases where there are SIP PBXs behind =

>>> unadministrated NATs or firewalls?
>>>
>>> Wouldn't we then need keepalives for SMTP as well, or how has the =

>>> e-mail infrastructure managed to solve this problem?
>>>
>>> Markus
>>>
>>>     =

>>>       =

>> _______________________________________________
>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use =

>> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

>>     =

>
>   =

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


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Thread-Topic: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use case
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From: "Sanjay Sinha (sanjsinh)" <sanjsinh@cisco.com>
To: "Jeroen van Bemmel" <jbemmel@zonnet.nl>,
	"Christer Holmberg" <christer.holmberg@ericsson.com>
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Cc: sip@ietf.org, john.elwell@siemens.com, drage@alcatel-lucent.com,
	Markus.Isomaki@nokia.com
Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use case
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In the asymmetric case, if the registration is established with a long dura=
tion, and for some reason if IP-PBX crashes before the registration expires=
, then SP n/w will not know about it until the registration expires, or it =
tries to send a request to that PBX.
If the registration interval is short, then this becomes similar to using s=
hort registration intervals to keep nat bindings open.

Sanjay

>-----Original Message-----
>From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On =

>Behalf Of Jeroen van Bemmel
>Sent: Wednesday, June 25, 2008 5:56 PM
>To: Christer Holmberg
>Cc: sip@ietf.org; drage@alcatel-lucent.com; =

>john.elwell@siemens.com; Markus.Isomaki@nokia.com
>Subject: Re: [Sip] Progress draft-holmberg-sip-keep: =

>proxy-to-proxy use case
>
>No, not a B2BUA. Each element would have a virtual UA function =

>"in parallel" to its regular function (e.g. being a proxy). =

>And indeed, one way to implement this would be to keep either =

>one (one way) or two (opposite ways) flows active between them.
>
>Like so:
>
>|-----------|                     |-----------|
>|   Proxy     |                     |  Proxy     |
>|-----------|                     |-----------|
>|     UA      | <---------->|      UA     |
>|-----------|                     |-----------|
>
>Each "UA" would implement both a simple registrar and a UAC =

>performing registration. It could also be setup =

>asymmetrically, with the (smaller) IP-PBX doing a single =

>registration towards the (bigger) Service provider network. =

>Registration expiry would denote a loss of connectivity.
>
>Regards,
>Jeroen
>
>
>Christer Holmberg wrote:
>> Hi,
>>
>> So, you are proposing that each element should be a B2BUA, =

>and both elements then register towards each other and use Outbound???
>>
>> Regards,
>>
>> Christer
>>
>> -----Original Message-----
>> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
>> Sent: 25. kes=E4kuuta 2008 22:57
>> To: Hadriel Kaplan
>> Cc: Markus.Isomaki@nokia.com; john.elwell@siemens.com; =

>> drage@alcatel-lucent.com; sip@ietf.org; Christer Holmberg
>> Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy =

>> use case
>>
>> Hadriel, Markus,
>>
>> Instead of standardizing keep-alives between proxies, how about we =

>> define a "virtual UA" on each element (similar to the one =

>described in
>> RFC3261 section 16.7 point 6) to be used to provide this =

>functionality? =

>> (using existing outbound functionality, perhaps both ways)
>>
>> Regards,
>> Jeroen
>>
>> Hadriel Kaplan wrote:
>>   =

>>> Yes I am of that same opinion - that any real "IP-PBX" or whatever =

>>> big enough NOT to be doing Registration, and to instead do =

>static provisioning or DNS, would be given a static hole/DMZ =

>address in their firewall/NAT.  But some of my customers have =

>told me otherwise. (interestingly mostly in APAC region) =

>There's also some concern that while a static entry is there =

>for inbound TCP connections, the PBX creates outbound ones to =

>the service provider which are ephemeral port sources and need =

>to live for very long durations (though why they can't just do =

>TCP keepalive is beyond me, but I'm no expert).
>>>
>>> But anyway, the big issue we've seen is that we need both =

>the PBX and the service provider box to detect failure before =

>an active call/request attempt is made; to trigger alternate =

>route selection without waiting for transport failure, and as =

>a method to detect liveness again and revert.  Today that's =

>almost exclusively done with Options requests as far as I've =

>seen, and lots of people don't seem to like that.
>>>
>>> -hadriel
>>>
>>>
>>>   =

>>>     =

>>>> -----Original Message-----
>>>> From: Markus.Isomaki@nokia.com [mailto:Markus.Isomaki@nokia.com]
>>>> Sent: Wednesday, June 25, 2008 3:11 PM
>>>> To: john.elwell@siemens.com; Hadriel Kaplan; =

>>>> drage@alcatel-lucent.com; sip@ietf.org
>>>> Cc: christer.holmberg@ericsson.com
>>>> Subject: RE: [Sip] Progress draft-holmberg-sip-keep: =

>proxy-to-proxy =

>>>> use case
>>>>
>>>> Hi,
>>>>
>>>> I'm a bit sceptical about the need for keep-alives between =

>proxies. =

>>>> It is of course entirely possible that an enterprise PBX is =

>>>> connected to (or peering with) a service provider proxy through a =

>>>> NAT and/or a firewall. However, wouldn't such a NAT or firewall be =

>>>> under the administration of either the enterprise itself =

>or its ISP =

>>>> (who quite often would be the SIP service provider), and the =

>>>> required port forwardings or firewall rules could be set =

>through administration.
>>>> This means that there would not be need for keepalive traffic to =

>>>> implicitely keep the mapping/pinhole open.
>>>>
>>>> Or are there really deployment cases where there are SIP =

>PBXs behind =

>>>> unadministrated NATs or firewalls?
>>>>
>>>> Wouldn't we then need keepalives for SMTP as well, or how has the =

>>>> e-mail infrastructure managed to solve this problem?
>>>>
>>>> Markus
>>>>
>>>>     =

>>>>       =

>>> _______________________________________________
>>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol Use =

>>> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

>>>     =

>>
>>   =

>_______________________________________________
>Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol Use =

>sip-implementors@cs.columbia.edu for questions on current sip =

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


From sip-bounces@ietf.org  Wed Jun 25 22:14:52 2008
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Subject: Re: [Sip] Second draft, RETIRE Charter
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On Jun 25, 2008, at 10:00 PM, Richard Shockey wrote:

> Yea .. RETIRE is the right name since you guys are all going to RETIRE
> before you get any consensus on what the solution is.

The first SIP SAML draft is getting close to pub-ready, and I think we  
can fix RFC 4474 with a small and discrete set of changes that have  
already been proposed in a draft. It is very close to right, just a  
little bit off because of Badness in our Networks.

>
> Or you are all going to RETIRE to Geneva to figure this out etc...


One of our dear friends has suggested to me (rather frequently) that  
all SIP related work should be ceded to the ITU, as it is now clearly  
a carrier-centric telephony protocol based on the assumption of a  
"smart network" and "crippled endpoints". Besides, Geneva is nice this  
time of year.

--
Dean
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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To: Jeroen van Bemmel <jbemmel@zonnet.nl>, Christer Holmberg
	<christer.holmberg@ericsson.com>
Date: Thu, 26 Jun 2008 01:56:37 -0400
Thread-Topic: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use case
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Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use case
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Hey Jeroen,
I'm not sure I follow you.  The SIP Forum SIP-Connect profile, and now TISP=
AN, make use of a REGISTER for an IP-PBX to register a whole set of AoR's t=
o be routed to.  For those devices that need that ability, it makes sense t=
o use outbound of course, and its inherent keepalive, regardless of being b=
ehind a NAT or not.  And I don't think Christer's suggesting otherwise.

But for IP-PBX's or proxies which do not themselves represent an AoR target=
 of requests, or do not represent one in the domain of the device they want=
 to perform keepalive with, why would we want to make them add such registr=
ation logic?  What would be the gain over just simply sending OPTIONS at th=
at point?

The beauty of a STUN or double-CRLF keepalive is, in my mind: it's trivial =
to construct, trivial to parse, very small and fixed size, can be separated=
 or handled at a lower layer, explicit in its use, and does not get stopped=
 by a SIP-layer overload control.  In short: it's a transport connection-la=
yer keepalive, no more no less.  And indicating it in the Via keeps it a ho=
p-link thing, backwards-compatible, little or no provisioning, and with no =
URI target addressing issues.

-hadriel

> -----Original Message-----
> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
> Sent: Wednesday, June 25, 2008 5:56 PM
> To: Christer Holmberg
> Cc: Hadriel Kaplan; Markus.Isomaki@nokia.com; john.elwell@siemens.com;
> drage@alcatel-lucent.com; sip@ietf.org
> Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use
> case
>
> No, not a B2BUA. Each element would have a virtual UA function "in
> parallel" to its regular function (e.g. being a proxy). And indeed, one
> way to implement this would be to keep either one (one way) or two
> (opposite ways) flows active between them.
>
> Like so:
>
> |-----------|                     |-----------|
> |   Proxy     |                     |  Proxy     |
> |-----------|                     |-----------|
> |     UA      | <---------->|      UA     |
> |-----------|                     |-----------|
>
> Each "UA" would implement both a simple registrar and a UAC performing
> registration. It could also be setup asymmetrically, with the (smaller)
> IP-PBX doing a single registration towards the (bigger) Service provider
> network. Registration expiry would denote a loss of connectivity.
>
> Regards,
> Jeroen
>
>
> Christer Holmberg wrote:
> > Hi,
> >
> > So, you are proposing that each element should be a B2BUA, and both
> elements then register towards each other and use Outbound???
> >
> > Regards,
> >
> > Christer
> >
> > -----Original Message-----
> > From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
> > Sent: 25. kes=E4kuuta 2008 22:57
> > To: Hadriel Kaplan
> > Cc: Markus.Isomaki@nokia.com; john.elwell@siemens.com; drage@alcatel-
> lucent.com; sip@ietf.org; Christer Holmberg
> > Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use
> case
> >
> > Hadriel, Markus,
> >
> > Instead of standardizing keep-alives between proxies, how about we
> define a "virtual UA" on each element (similar to the one described in
> > RFC3261 section 16.7 point 6) to be used to provide this functionality?
> > (using existing outbound functionality, perhaps both ways)
> >
> > Regards,
> > Jeroen
> >
> > Hadriel Kaplan wrote:
> >
> >> Yes I am of that same opinion - that any real "IP-PBX" or whatever big
> >> enough NOT to be doing Registration, and to instead do static
> provisioning or DNS, would be given a static hole/DMZ address in their
> firewall/NAT.  But some of my customers have told me otherwise.
> (interestingly mostly in APAC region) There's also some concern that while
> a static entry is there for inbound TCP connections, the PBX creates
> outbound ones to the service provider which are ephemeral port sources and
> need to live for very long durations (though why they can't just do TCP
> keepalive is beyond me, but I'm no expert).
> >>
> >> But anyway, the big issue we've seen is that we need both the PBX and
> the service provider box to detect failure before an active call/request
> attempt is made; to trigger alternate route selection without waiting for
> transport failure, and as a method to detect liveness again and revert.
> Today that's almost exclusively done with Options requests as far as I've
> seen, and lots of people don't seem to like that.
> >>
> >> -hadriel
> >>
> >>
> >>
> >>
> >>> -----Original Message-----
> >>> From: Markus.Isomaki@nokia.com [mailto:Markus.Isomaki@nokia.com]
> >>> Sent: Wednesday, June 25, 2008 3:11 PM
> >>> To: john.elwell@siemens.com; Hadriel Kaplan;
> >>> drage@alcatel-lucent.com; sip@ietf.org
> >>> Cc: christer.holmberg@ericsson.com
> >>> Subject: RE: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy
> >>> use case
> >>>
> >>> Hi,
> >>>
> >>> I'm a bit sceptical about the need for keep-alives between proxies.
> >>> It is of course entirely possible that an enterprise PBX is connected
> >>> to (or peering with) a service provider proxy through a NAT and/or a
> >>> firewall. However, wouldn't such a NAT or firewall be under the
> >>> administration of either the enterprise itself or its ISP (who quite
> >>> often would be the SIP service provider), and the required port
> >>> forwardings or firewall rules could be set through administration.
> >>> This means that there would not be need for keepalive traffic to
> >>> implicitely keep the mapping/pinhole open.
> >>>
> >>> Or are there really deployment cases where there are SIP PBXs behind
> >>> unadministrated NATs or firewalls?
> >>>
> >>> Wouldn't we then need keepalives for SMTP as well, or how has the
> >>> e-mail infrastructure managed to solve this problem?
> >>>
> >>> Markus
> >>>
> >>>
> >>>
> >> _______________________________________________
> >> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> >> This list is for NEW development of the core SIP Protocol Use
> >> sip-implementors@cs.columbia.edu for questions on current sip Use
> >> sipping@ietf.org for new developments on the application of sip
> >>
> >>
> >>
> >
> >
_______________________________________________
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Use sip-implementors@cs.columbia.edu for questions on current sip
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From: "Sudhir Kumar Reddy" <t.sudhirkumar@gmail.com>
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Subject: [Sip] can SDP contain 2 'M' lines
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Hi All,

In SDP can we have two 'M' lines for re-INVITE i.e. for call hold.

if yes what is RFC / draft

Any response is highly appreciated

Regards
Sudhir

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Hi All,<br><br>In SDP can we have two &#39;M&#39; lines for re-INVITE i.e. for call hold.<br><br>if yes what is RFC / draft<br><br>Any response is highly appreciated<br><br>Regards<br>Sudhir

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From: "Fries, Steffen (CT)" <steffen.fries@siemens.com>
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Subject: Re: [Sip] Second draft, RETIRE Charter
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Hi Dean,

the revised charter sounds good. 
I'm not sure I like the agronym as it somehow sounds that it will take
longer to solve that problem. 
But tht is just naming, the content is more important ;-)

Ciao
	Steffen

PS: How about RISE: Roles and Identities in SIP Environments?

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of Dean Willis
> Sent: Wednesday, June 25, 2008 6:13 PM
> To: sip@ietf.org
> Subject: [Sip] Second draft, RETIRE Charter
> 
> I've revised the proposed charter fora RETIRE working group 
> based on list comments. This is still a very early concept, 
> and the formation of such a working group is contingent on a 
> great many things that are outside of my control.
> 
> 
> --------
> 
> 
> The "REal Time Identity and  Role Expression" (RETIRE) 
> Working Group's purpose is to standardize the expression and 
> assertion of identity and roles needed for real-time 
> communications with the IETF's Realtime Applications 
> Infrastructure (RAI) area protocols, primarily the Session 
> Initiation Protocol (SIP) and others used in conjunction with SIP.
> 
> By "Identity", we mean authenticated addresses of record (as 
> defined in RFC 4474 "Enhancements for Authenticated Identity 
> Management in the Session Initiation Protocol". RFC 4474 
> provides a means by which a SIP server may vouch for the 
> "From" address of a request transiting that server. It 
> further provides a cryptographic mechanism for testing the 
> source of the assertion. In the typical case, this mechanism 
> is dependent on the same Public Key Infrastructure that 
> supports secure HTTP services. Since RFC 4474 provides only 
> authentication of the source of SIP request messages and it 
> is also important to determine the identity of the sender of 
> responses, RFC 4916 "Connected Identity in the Session 
> Initiation Protocol" provides authenticated identity in an 
> ongoing SIP dialog by applying RFC 4474 to requests in the 
> reverse direction. While adequate for some scenarios, this 
> mechanism is not useful for SIP interactions having a single 
> non-dialog transaction, such as MESSAGE requests. RFC 4474 
> suffers additional limitations relating to its use with E.164 
> numbers, gateways, session border controllers, and 
> back-to-back user agents.
> 
> By "Role", we mean aspects of a a system user that are 
> related identity, but more dynamic in nature. For example, 
> Alice might delegate her calls to Bob. A calling party, 
> expecting to reach Alice but reaching Bob, needs to know that 
> this is a reasonable thing and what Bob's role in relation to 
> Alice is. This problem is in general referred to as the 
> "unanticipated respondent problem", and it has impacts both 
> at the level of user expectations and key management.  
> Proposals for fixing it have included deprecation of proxy 
> retargeting and the use of a new SIP provisional response to 
> inform the calling party about the retargeting operation.
> 
> Security Assertion Markup Language (SAML) is an XML-based 
> standard for exchanging authentication and authorization data 
> between security domains, that is, between an identity 
> provider (a producer of
> assertions) and a service provider (a consumer of 
> assertions). SAML is a product of the OASIS Security Services 
> Technical Committee. The SIP working group worked for several 
> years on applying SAML to SIP with mixed results, but the 
> technique appears to have great promise.
> 
> 
> The initial deliverables of the RETIRE working group are:
> 
> 1) A standards-track revision of RFC 4474 that resolves its 
> limitations with respect to E.164 numbers, gateways, session 
> border controllers,  and back-to-back user agents. This 
> revision will also address the use of Authenticated Identity 
> Management in conjunction with SIP response messages. It may 
> also be necessary to document the impact of such changes, if 
> any, on the DTLS-SRTP framework.
> 
> 2) Specify a solution (or set of solutions)  for the 
> Unanticipated Respondent Problem in SIP.
> 
> 3) Guidelines for the use of E.164 numbers as SIP identifiers.
> 
> 4) Guidelines for the use of SAML with SIP for the purposes 
> of expressing identity and role, specifically in the contexts 
> of #1, #2 and #3 above.
> 
> 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use 
> sip-implementors@cs.columbia.edu for questions on current sip 
> Use sipping@ietf.org for new developments on the application of sip
> 
_______________________________________________
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From: Jeroen van Bemmel <jbemmel@zonnet.nl>
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To: Hadriel Kaplan <HKaplan@acmepacket.com>
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Cc: "sip@ietf.org" <sip@ietf.org>,
	"drage@alcatel-lucent.com" <drage@alcatel-lucent.com>,
	"Markus.Isomaki@nokia.com" <Markus.Isomaki@nokia.com>,
	"john.elwell@siemens.com" <john.elwell@siemens.com>,
	Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use case
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Hadriel,

I did not intend to suggest to use the registration interval timer for =

keep-alive, of course all the regular flow keep-alive mechanisms of =

outbound could be used.

The benefit is that we would reuse the efforts put in outbound, instead =

of inventing yet another, only marginally different, mechanism for each =

specific case. From the sending element point of view, sending a =

REGISTER isn't very different from sending an OPTIONS; subsequent =

keep-alives can then become STUN or double CRLF, until the registration =

needs to be refreshed.

Using REGISTER instead of OPTIONS also provides the means to use the =

standard authentication mechanisms, besides reusing outbound (and =

there's a few other reusable things, like Path headers, but let's keep =

it simple for the moment)

Regards,
Jeroen

Hadriel Kaplan wrote:
> Hey Jeroen,
> I'm not sure I follow you.  The SIP Forum SIP-Connect profile, and now TI=
SPAN, make use of a REGISTER for an IP-PBX to register a whole set of AoR's=
 to be routed to.  For those devices that need that ability, it makes sense=
 to use outbound of course, and its inherent keepalive, regardless of being=
 behind a NAT or not.  And I don't think Christer's suggesting otherwise.
>
> But for IP-PBX's or proxies which do not themselves represent an AoR targ=
et of requests, or do not represent one in the domain of the device they wa=
nt to perform keepalive with, why would we want to make them add such regis=
tration logic?  What would be the gain over just simply sending OPTIONS at =
that point?
>
> The beauty of a STUN or double-CRLF keepalive is, in my mind: it's trivia=
l to construct, trivial to parse, very small and fixed size, can be separat=
ed or handled at a lower layer, explicit in its use, and does not get stopp=
ed by a SIP-layer overload control.  In short: it's a transport connection-=
layer keepalive, no more no less.  And indicating it in the Via keeps it a =
hop-link thing, backwards-compatible, little or no provisioning, and with n=
o URI target addressing issues.
>
> -hadriel
>
>   =

>> -----Original Message-----
>> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
>> Sent: Wednesday, June 25, 2008 5:56 PM
>> To: Christer Holmberg
>> Cc: Hadriel Kaplan; Markus.Isomaki@nokia.com; john.elwell@siemens.com;
>> drage@alcatel-lucent.com; sip@ietf.org
>> Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use
>> case
>>
>> No, not a B2BUA. Each element would have a virtual UA function "in
>> parallel" to its regular function (e.g. being a proxy). And indeed, one
>> way to implement this would be to keep either one (one way) or two
>> (opposite ways) flows active between them.
>>
>> Like so:
>>
>> |-----------|                     |-----------|
>> |   Proxy     |                     |  Proxy     |
>> |-----------|                     |-----------|
>> |     UA      | <---------->|      UA     |
>> |-----------|                     |-----------|
>>
>> Each "UA" would implement both a simple registrar and a UAC performing
>> registration. It could also be setup asymmetrically, with the (smaller)
>> IP-PBX doing a single registration towards the (bigger) Service provider
>> network. Registration expiry would denote a loss of connectivity.
>>
>> Regards,
>> Jeroen
>>
>>
>> Christer Holmberg wrote:
>>     =

>>> Hi,
>>>
>>> So, you are proposing that each element should be a B2BUA, and both
>>>       =

>> elements then register towards each other and use Outbound???
>>     =

>>> Regards,
>>>
>>> Christer
>>>
>>> -----Original Message-----
>>> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
>>> Sent: 25. kes=E4kuuta 2008 22:57
>>> To: Hadriel Kaplan
>>> Cc: Markus.Isomaki@nokia.com; john.elwell@siemens.com; drage@alcatel-
>>>       =

>> lucent.com; sip@ietf.org; Christer Holmberg
>>     =

>>> Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use
>>>       =

>> case
>>     =

>>> Hadriel, Markus,
>>>
>>> Instead of standardizing keep-alives between proxies, how about we
>>>       =

>> define a "virtual UA" on each element (similar to the one described in
>>     =

>>> RFC3261 section 16.7 point 6) to be used to provide this functionality?
>>> (using existing outbound functionality, perhaps both ways)
>>>
>>> Regards,
>>> Jeroen
>>>
>>> Hadriel Kaplan wrote:
>>>
>>>       =

>>>> Yes I am of that same opinion - that any real "IP-PBX" or whatever big
>>>> enough NOT to be doing Registration, and to instead do static
>>>>         =

>> provisioning or DNS, would be given a static hole/DMZ address in their
>> firewall/NAT.  But some of my customers have told me otherwise.
>> (interestingly mostly in APAC region) There's also some concern that whi=
le
>> a static entry is there for inbound TCP connections, the PBX creates
>> outbound ones to the service provider which are ephemeral port sources a=
nd
>> need to live for very long durations (though why they can't just do TCP
>> keepalive is beyond me, but I'm no expert).
>>     =

>>>> But anyway, the big issue we've seen is that we need both the PBX and
>>>>         =

>> the service provider box to detect failure before an active call/request
>> attempt is made; to trigger alternate route selection without waiting for
>> transport failure, and as a method to detect liveness again and revert.
>> Today that's almost exclusively done with Options requests as far as I've
>> seen, and lots of people don't seem to like that.
>>     =

>>>> -hadriel
>>>>
>>>>
>>>>
>>>>
>>>>         =

>>>>> -----Original Message-----
>>>>> From: Markus.Isomaki@nokia.com [mailto:Markus.Isomaki@nokia.com]
>>>>> Sent: Wednesday, June 25, 2008 3:11 PM
>>>>> To: john.elwell@siemens.com; Hadriel Kaplan;
>>>>> drage@alcatel-lucent.com; sip@ietf.org
>>>>> Cc: christer.holmberg@ericsson.com
>>>>> Subject: RE: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy
>>>>> use case
>>>>>
>>>>> Hi,
>>>>>
>>>>> I'm a bit sceptical about the need for keep-alives between proxies.
>>>>> It is of course entirely possible that an enterprise PBX is connected
>>>>> to (or peering with) a service provider proxy through a NAT and/or a
>>>>> firewall. However, wouldn't such a NAT or firewall be under the
>>>>> administration of either the enterprise itself or its ISP (who quite
>>>>> often would be the SIP service provider), and the required port
>>>>> forwardings or firewall rules could be set through administration.
>>>>> This means that there would not be need for keepalive traffic to
>>>>> implicitely keep the mapping/pinhole open.
>>>>>
>>>>> Or are there really deployment cases where there are SIP PBXs behind
>>>>> unadministrated NATs or firewalls?
>>>>>
>>>>> Wouldn't we then need keepalives for SMTP as well, or how has the
>>>>> e-mail infrastructure managed to solve this problem?
>>>>>
>>>>> Markus
>>>>>
>>>>>
>>>>>
>>>>>           =

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

>>>       =

>
>   =

_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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Date: Thu, 26 Jun 2008 08:03:34 +0100
From: "Elwell, John" <john.elwell@siemens.com>
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Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use case
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Jeroen,

Yes, this is effectively what you have when an IPPBX is required to registe=
r with a service provider. But in that case, there might also be a need to =
implement sip-outbound if there are unadministered NATs/firewalls in the wa=
y. Indeed, sip-outbound would operate between a virtual UA on the IPPBX sid=
e and a registrar on the service provider side.

What I am interested in gaining from this discussion is whether there are p=
roxy-to-proxy cases where sip-outbound is not appropriate but keep-alive (o=
r heartbeat) is needed.

John

> -----Original Message-----
> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl] =

> Sent: 25 June 2008 22:56
> To: Christer Holmberg
> Cc: Hadriel Kaplan; Markus.Isomaki@nokia.com; Elwell, John; =

> drage@alcatel-lucent.com; sip@ietf.org
> Subject: Re: [Sip] Progress draft-holmberg-sip-keep: =

> proxy-to-proxy use case
> =

> No, not a B2BUA. Each element would have a virtual UA function "in =

> parallel" to its regular function (e.g. being a proxy). And =

> indeed, one =

> way to implement this would be to keep either one (one way) or two =

> (opposite ways) flows active between them.
> =

> Like so:
> =

> |-----------|                     |-----------|
> |   Proxy     |                     |  Proxy     |
> |-----------|                     |-----------|
> |     UA      | <---------->|      UA     |
> |-----------|                     |-----------|
> =

> Each "UA" would implement both a simple registrar and a UAC =

> performing =

> registration. It could also be setup asymmetrically, with the =

> (smaller) =

> IP-PBX doing a single registration towards the (bigger) =

> Service provider =

> network. Registration expiry would denote a loss of connectivity.
> =

> Regards,
> Jeroen
> =

> =

> Christer Holmberg wrote:
> > Hi,
> >
> > So, you are proposing that each element should be a B2BUA, =

> and both elements then register towards each other and use Outbound???
> >
> > Regards,
> >
> > Christer =

> >
> > -----Original Message-----
> > From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl] =

> > Sent: 25. kes=E4kuuta 2008 22:57
> > To: Hadriel Kaplan
> > Cc: Markus.Isomaki@nokia.com; john.elwell@siemens.com; =

> drage@alcatel-lucent.com; sip@ietf.org; Christer Holmberg
> > Subject: Re: [Sip] Progress draft-holmberg-sip-keep: =

> proxy-to-proxy use case
> >
> > Hadriel, Markus,
> >
> > Instead of standardizing keep-alives between proxies, how =

> about we define a "virtual UA" on each element (similar to =

> the one described in
> > RFC3261 section 16.7 point 6) to be used to provide this =

> functionality? =

> > (using existing outbound functionality, perhaps both ways)
> >
> > Regards,
> > Jeroen
> >
> > Hadriel Kaplan wrote:
> >   =

> >> Yes I am of that same opinion - that any real "IP-PBX" or =

> whatever big =

> >> enough NOT to be doing Registration, and to instead do =

> static provisioning or DNS, would be given a static hole/DMZ =

> address in their firewall/NAT.  But some of my customers have =

> told me otherwise. (interestingly mostly in APAC region) =

> There's also some concern that while a static entry is there =

> for inbound TCP connections, the PBX creates outbound ones to =

> the service provider which are ephemeral port sources and =

> need to live for very long durations (though why they can't =

> just do TCP keepalive is beyond me, but I'm no expert).
> >>
> >> But anyway, the big issue we've seen is that we need both =

> the PBX and the service provider box to detect failure before =

> an active call/request attempt is made; to trigger alternate =

> route selection without waiting for transport failure, and as =

> a method to detect liveness again and revert.  Today that's =

> almost exclusively done with Options requests as far as I've =

> seen, and lots of people don't seem to like that.
> >>
> >> -hadriel
> >>
> >>
> >>   =

> >>     =

> >>> -----Original Message-----
> >>> From: Markus.Isomaki@nokia.com [mailto:Markus.Isomaki@nokia.com]
> >>> Sent: Wednesday, June 25, 2008 3:11 PM
> >>> To: john.elwell@siemens.com; Hadriel Kaplan; =

> >>> drage@alcatel-lucent.com; sip@ietf.org
> >>> Cc: christer.holmberg@ericsson.com
> >>> Subject: RE: [Sip] Progress draft-holmberg-sip-keep: =

> proxy-to-proxy =

> >>> use case
> >>>
> >>> Hi,
> >>>
> >>> I'm a bit sceptical about the need for keep-alives =

> between proxies. =

> >>> It is of course entirely possible that an enterprise PBX =

> is connected =

> >>> to (or peering with) a service provider proxy through a =

> NAT and/or a =

> >>> firewall. However, wouldn't such a NAT or firewall be under the =

> >>> administration of either the enterprise itself or its ISP =

> (who quite =

> >>> often would be the SIP service provider), and the required port =

> >>> forwardings or firewall rules could be set through =

> administration. =

> >>> This means that there would not be need for keepalive traffic to =

> >>> implicitely keep the mapping/pinhole open.
> >>>
> >>> Or are there really deployment cases where there are SIP =

> PBXs behind =

> >>> unadministrated NATs or firewalls?
> >>>
> >>> Wouldn't we then need keepalives for SMTP as well, or how has the =

> >>> e-mail infrastructure managed to solve this problem?
> >>>
> >>> Markus
> >>>
> >>>     =

> >>>       =

> >> _______________________________________________
> >> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> >> This list is for NEW development of the core SIP Protocol Use =

> >> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

> >>     =

> >
> >   =

> =

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


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This is what sip-outbound is for.

John =


> -----Original Message-----
> From: Sanjay Sinha (sanjsinh) [mailto:sanjsinh@cisco.com] =

> Sent: 26 June 2008 05:03
> To: Jeroen van Bemmel; Christer Holmberg
> Cc: sip@ietf.org; drage@alcatel-lucent.com; Elwell, John; =

> Markus.Isomaki@nokia.com
> Subject: RE: [Sip] Progress draft-holmberg-sip-keep: =

> proxy-to-proxy use case
> =

> In the asymmetric case, if the registration is established =

> with a long duration, and for some reason if IP-PBX crashes =

> before the registration expires, then SP n/w will not know =

> about it until the registration expires, or it tries to send =

> a request to that PBX.
> If the registration interval is short, then this becomes =

> similar to using short registration intervals to keep nat =

> bindings open.
> =

> Sanjay
> =

> >-----Original Message-----
> >From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On =

> >Behalf Of Jeroen van Bemmel
> >Sent: Wednesday, June 25, 2008 5:56 PM
> >To: Christer Holmberg
> >Cc: sip@ietf.org; drage@alcatel-lucent.com; =

> >john.elwell@siemens.com; Markus.Isomaki@nokia.com
> >Subject: Re: [Sip] Progress draft-holmberg-sip-keep: =

> >proxy-to-proxy use case
> >
> >No, not a B2BUA. Each element would have a virtual UA function =

> >"in parallel" to its regular function (e.g. being a proxy). =

> >And indeed, one way to implement this would be to keep either =

> >one (one way) or two (opposite ways) flows active between them.
> >
> >Like so:
> >
> >|-----------|                     |-----------|
> >|   Proxy     |                     |  Proxy     |
> >|-----------|                     |-----------|
> >|     UA      | <---------->|      UA     |
> >|-----------|                     |-----------|
> >
> >Each "UA" would implement both a simple registrar and a UAC =

> >performing registration. It could also be setup =

> >asymmetrically, with the (smaller) IP-PBX doing a single =

> >registration towards the (bigger) Service provider network. =

> >Registration expiry would denote a loss of connectivity.
> >
> >Regards,
> >Jeroen
> >
> >
> >Christer Holmberg wrote:
> >> Hi,
> >>
> >> So, you are proposing that each element should be a B2BUA, =

> >and both elements then register towards each other and use =

> Outbound???
> >>
> >> Regards,
> >>
> >> Christer
> >>
> >> -----Original Message-----
> >> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
> >> Sent: 25. kes=E4kuuta 2008 22:57
> >> To: Hadriel Kaplan
> >> Cc: Markus.Isomaki@nokia.com; john.elwell@siemens.com; =

> >> drage@alcatel-lucent.com; sip@ietf.org; Christer Holmberg
> >> Subject: Re: [Sip] Progress draft-holmberg-sip-keep: =

> proxy-to-proxy =

> >> use case
> >>
> >> Hadriel, Markus,
> >>
> >> Instead of standardizing keep-alives between proxies, how about we =

> >> define a "virtual UA" on each element (similar to the one =

> >described in
> >> RFC3261 section 16.7 point 6) to be used to provide this =

> >functionality? =

> >> (using existing outbound functionality, perhaps both ways)
> >>
> >> Regards,
> >> Jeroen
> >>
> >> Hadriel Kaplan wrote:
> >>   =

> >>> Yes I am of that same opinion - that any real "IP-PBX" or =

> whatever =

> >>> big enough NOT to be doing Registration, and to instead do =

> >static provisioning or DNS, would be given a static hole/DMZ =

> >address in their firewall/NAT.  But some of my customers have =

> >told me otherwise. (interestingly mostly in APAC region) =

> >There's also some concern that while a static entry is there =

> >for inbound TCP connections, the PBX creates outbound ones to =

> >the service provider which are ephemeral port sources and need =

> >to live for very long durations (though why they can't just do =

> >TCP keepalive is beyond me, but I'm no expert).
> >>>
> >>> But anyway, the big issue we've seen is that we need both =

> >the PBX and the service provider box to detect failure before =

> >an active call/request attempt is made; to trigger alternate =

> >route selection without waiting for transport failure, and as =

> >a method to detect liveness again and revert.  Today that's =

> >almost exclusively done with Options requests as far as I've =

> >seen, and lots of people don't seem to like that.
> >>>
> >>> -hadriel
> >>>
> >>>
> >>>   =

> >>>     =

> >>>> -----Original Message-----
> >>>> From: Markus.Isomaki@nokia.com [mailto:Markus.Isomaki@nokia.com]
> >>>> Sent: Wednesday, June 25, 2008 3:11 PM
> >>>> To: john.elwell@siemens.com; Hadriel Kaplan; =

> >>>> drage@alcatel-lucent.com; sip@ietf.org
> >>>> Cc: christer.holmberg@ericsson.com
> >>>> Subject: RE: [Sip] Progress draft-holmberg-sip-keep: =

> >proxy-to-proxy =

> >>>> use case
> >>>>
> >>>> Hi,
> >>>>
> >>>> I'm a bit sceptical about the need for keep-alives between =

> >proxies. =

> >>>> It is of course entirely possible that an enterprise PBX is =

> >>>> connected to (or peering with) a service provider proxy =

> through a =

> >>>> NAT and/or a firewall. However, wouldn't such a NAT or =

> firewall be =

> >>>> under the administration of either the enterprise itself =

> >or its ISP =

> >>>> (who quite often would be the SIP service provider), and the =

> >>>> required port forwardings or firewall rules could be set =

> >through administration.
> >>>> This means that there would not be need for keepalive traffic to =

> >>>> implicitely keep the mapping/pinhole open.
> >>>>
> >>>> Or are there really deployment cases where there are SIP =

> >PBXs behind =

> >>>> unadministrated NATs or firewalls?
> >>>>
> >>>> Wouldn't we then need keepalives for SMTP as well, or =

> how has the =

> >>>> e-mail infrastructure managed to solve this problem?
> >>>>
> >>>> Markus
> >>>>
> >>>>     =

> >>>>       =

> >>> _______________________________________________
> >>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> >>> This list is for NEW development of the core SIP Protocol Use =

> >>> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

> >>>     =

> >>
> >>   =

> >_______________________________________________
> >Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> >This list is for NEW development of the core SIP Protocol Use =

> >sip-implementors@cs.columbia.edu for questions on current sip =

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

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


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Thread-Topic: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use case
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Jeroen van Bemmel" <jbemmel@zonnet.nl>,
	"Hadriel Kaplan" <HKaplan@acmepacket.com>
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	Markus.Isomaki@nokia.com
Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use case
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Jeroen,

I don't understand what benefit it would be to establish outbound registrat=
ions between all SIP entities that want to use the keep-alive/heartbeat. Ye=
s, we would "re-use an existing mechanism", but in my opinion it would real=
ly be an overkill,a hack - and anything but keep-it-simple...

Also, how would the proxies know which proxies it should establish registra=
tions which?

Also, it wouldn't work in the use-case with UEs that don't register.

Regards,

Christer


-----Original Message-----
From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl] =

Sent: 26. kes=E4kuuta 2008 9:50
To: Hadriel Kaplan
Cc: Christer Holmberg; Markus.Isomaki@nokia.com; john.elwell@siemens.com; d=
rage@alcatel-lucent.com; sip@ietf.org
Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use case

Hadriel,

I did not intend to suggest to use the registration interval timer for keep=
-alive, of course all the regular flow keep-alive mechanisms of outbound co=
uld be used.

The benefit is that we would reuse the efforts put in outbound, instead of =
inventing yet another, only marginally different, mechanism for each specif=
ic case. From the sending element point of view, sending a REGISTER isn't v=
ery different from sending an OPTIONS; subsequent keep-alives can then beco=
me STUN or double CRLF, until the registration needs to be refreshed.

Using REGISTER instead of OPTIONS also provides the means to use the standa=
rd authentication mechanisms, besides reusing outbound (and there's a few o=
ther reusable things, like Path headers, but let's keep it simple for the m=
oment)

Regards,
Jeroen

Hadriel Kaplan wrote:
> Hey Jeroen,
> I'm not sure I follow you.  The SIP Forum SIP-Connect profile, and now TI=
SPAN, make use of a REGISTER for an IP-PBX to register a whole set of AoR's=
 to be routed to.  For those devices that need that ability, it makes sense=
 to use outbound of course, and its inherent keepalive, regardless of being=
 behind a NAT or not.  And I don't think Christer's suggesting otherwise.
>
> But for IP-PBX's or proxies which do not themselves represent an AoR targ=
et of requests, or do not represent one in the domain of the device they wa=
nt to perform keepalive with, why would we want to make them add such regis=
tration logic?  What would be the gain over just simply sending OPTIONS at =
that point?
>
> The beauty of a STUN or double-CRLF keepalive is, in my mind: it's trivia=
l to construct, trivial to parse, very small and fixed size, can be separat=
ed or handled at a lower layer, explicit in its use, and does not get stopp=
ed by a SIP-layer overload control.  In short: it's a transport connection-=
layer keepalive, no more no less.  And indicating it in the Via keeps it a =
hop-link thing, backwards-compatible, little or no provisioning, and with n=
o URI target addressing issues.
>
> -hadriel
>
>   =

>> -----Original Message-----
>> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
>> Sent: Wednesday, June 25, 2008 5:56 PM
>> To: Christer Holmberg
>> Cc: Hadriel Kaplan; Markus.Isomaki@nokia.com; =

>> john.elwell@siemens.com; drage@alcatel-lucent.com; sip@ietf.org
>> Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy =

>> use case
>>
>> No, not a B2BUA. Each element would have a virtual UA function "in =

>> parallel" to its regular function (e.g. being a proxy). And indeed, =

>> one way to implement this would be to keep either one (one way) or =

>> two (opposite ways) flows active between them.
>>
>> Like so:
>>
>> |-----------|                     |-----------|
>> |   Proxy     |                     |  Proxy     |
>> |-----------|                     |-----------|
>> |     UA      | <---------->|      UA     |
>> |-----------|                     |-----------|
>>
>> Each "UA" would implement both a simple registrar and a UAC =

>> performing registration. It could also be setup asymmetrically, with =

>> the (smaller) IP-PBX doing a single registration towards the (bigger) =

>> Service provider network. Registration expiry would denote a loss of con=
nectivity.
>>
>> Regards,
>> Jeroen
>>
>>
>> Christer Holmberg wrote:
>>     =

>>> Hi,
>>>
>>> So, you are proposing that each element should be a B2BUA, and both
>>>       =

>> elements then register towards each other and use Outbound???
>>     =

>>> Regards,
>>>
>>> Christer
>>>
>>> -----Original Message-----
>>> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
>>> Sent: 25. kes=E4kuuta 2008 22:57
>>> To: Hadriel Kaplan
>>> Cc: Markus.Isomaki@nokia.com; john.elwell@siemens.com; =

>>> drage@alcatel-
>>>       =

>> lucent.com; sip@ietf.org; Christer Holmberg
>>     =

>>> Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy =

>>> use
>>>       =

>> case
>>     =

>>> Hadriel, Markus,
>>>
>>> Instead of standardizing keep-alives between proxies, how about we
>>>       =

>> define a "virtual UA" on each element (similar to the one described =

>> in
>>     =

>>> RFC3261 section 16.7 point 6) to be used to provide this functionality?
>>> (using existing outbound functionality, perhaps both ways)
>>>
>>> Regards,
>>> Jeroen
>>>
>>> Hadriel Kaplan wrote:
>>>
>>>       =

>>>> Yes I am of that same opinion - that any real "IP-PBX" or whatever =

>>>> big enough NOT to be doing Registration, and to instead do static
>>>>         =

>> provisioning or DNS, would be given a static hole/DMZ address in =

>> their firewall/NAT.  But some of my customers have told me otherwise.
>> (interestingly mostly in APAC region) There's also some concern that =

>> while a static entry is there for inbound TCP connections, the PBX =

>> creates outbound ones to the service provider which are ephemeral =

>> port sources and need to live for very long durations (though why =

>> they can't just do TCP keepalive is beyond me, but I'm no expert).
>>     =

>>>> But anyway, the big issue we've seen is that we need both the PBX =

>>>> and
>>>>         =

>> the service provider box to detect failure before an active =

>> call/request attempt is made; to trigger alternate route selection =

>> without waiting for transport failure, and as a method to detect livenes=
s again and revert.
>> Today that's almost exclusively done with Options requests as far as =

>> I've seen, and lots of people don't seem to like that.
>>     =

>>>> -hadriel
>>>>
>>>>
>>>>
>>>>
>>>>         =

>>>>> -----Original Message-----
>>>>> From: Markus.Isomaki@nokia.com [mailto:Markus.Isomaki@nokia.com]
>>>>> Sent: Wednesday, June 25, 2008 3:11 PM
>>>>> To: john.elwell@siemens.com; Hadriel Kaplan; =

>>>>> drage@alcatel-lucent.com; sip@ietf.org
>>>>> Cc: christer.holmberg@ericsson.com
>>>>> Subject: RE: [Sip] Progress draft-holmberg-sip-keep: =

>>>>> proxy-to-proxy use case
>>>>>
>>>>> Hi,
>>>>>
>>>>> I'm a bit sceptical about the need for keep-alives between proxies.
>>>>> It is of course entirely possible that an enterprise PBX is =

>>>>> connected to (or peering with) a service provider proxy through a =

>>>>> NAT and/or a firewall. However, wouldn't such a NAT or firewall be =

>>>>> under the administration of either the enterprise itself or its =

>>>>> ISP (who quite often would be the SIP service provider), and the =

>>>>> required port forwardings or firewall rules could be set through admi=
nistration.
>>>>> This means that there would not be need for keepalive traffic to =

>>>>> implicitely keep the mapping/pinhole open.
>>>>>
>>>>> Or are there really deployment cases where there are SIP PBXs =

>>>>> behind unadministrated NATs or firewalls?
>>>>>
>>>>> Wouldn't we then need keepalives for SMTP as well, or how has the =

>>>>> e-mail infrastructure managed to solve this problem?
>>>>>
>>>>> Markus
>>>>>
>>>>>
>>>>>
>>>>>           =

>>>> _______________________________________________
>>>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>>>> This list is for NEW development of the core SIP Protocol Use =

>>>> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

>>>       =

>
>   =

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


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Cc: sip@ietf.org, drage@alcatel-lucent.com, john.elwell@siemens.com,
	Markus.Isomaki@nokia.com
Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use case
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Christer,

The only solution I can think of that would give less processing, would =

be to simply open a TCP connection (or UDP flow with STUN) without =

sending any request. Less "overkill", but it would open up a bunch of =

security issues ( e.g. TCP connection resources being consumed by =

unauthorized elements, leading to DoS; all the concerns with STUNning =

before checking support, etc. ). So I don't see how sending at least one =

SIP request could be avoided. REGISTER is as good as any.

Which proxies to register with: that would of course have to be =

provisioned somehow, this is no different from having to provision which =

outbound proxy to use

UEs that don't register: the idea is to make all such UEs register, so =

there'd be no such case left

Regards,
Jeroen

Christer Holmberg wrote:
> Jeroen,
>
> I don't understand what benefit it would be to establish outbound registr=
ations between all SIP entities that want to use the keep-alive/heartbeat. =
Yes, we would "re-use an existing mechanism", but in my opinion it would re=
ally be an overkill,a hack - and anything but keep-it-simple...
>
> Also, how would the proxies know which proxies it should establish regist=
rations which?
>
> Also, it wouldn't work in the use-case with UEs that don't register.
>
> Regards,
>
> Christer
>
>
> -----Original Message-----
> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl] =

> Sent: 26. kes=E4kuuta 2008 9:50
> To: Hadriel Kaplan
> Cc: Christer Holmberg; Markus.Isomaki@nokia.com; john.elwell@siemens.com;=
 drage@alcatel-lucent.com; sip@ietf.org
> Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use c=
ase
>
> Hadriel,
>
> I did not intend to suggest to use the registration interval timer for ke=
ep-alive, of course all the regular flow keep-alive mechanisms of outbound =
could be used.
>
> The benefit is that we would reuse the efforts put in outbound, instead o=
f inventing yet another, only marginally different, mechanism for each spec=
ific case. From the sending element point of view, sending a REGISTER isn't=
 very different from sending an OPTIONS; subsequent keep-alives can then be=
come STUN or double CRLF, until the registration needs to be refreshed.
>
> Using REGISTER instead of OPTIONS also provides the means to use the stan=
dard authentication mechanisms, besides reusing outbound (and there's a few=
 other reusable things, like Path headers, but let's keep it simple for the=
 moment)
>
> Regards,
> Jeroen
>
> Hadriel Kaplan wrote:
>   =

>> Hey Jeroen,
>> I'm not sure I follow you.  The SIP Forum SIP-Connect profile, and now T=
ISPAN, make use of a REGISTER for an IP-PBX to register a whole set of AoR'=
s to be routed to.  For those devices that need that ability, it makes sens=
e to use outbound of course, and its inherent keepalive, regardless of bein=
g behind a NAT or not.  And I don't think Christer's suggesting otherwise.
>>
>> But for IP-PBX's or proxies which do not themselves represent an AoR tar=
get of requests, or do not represent one in the domain of the device they w=
ant to perform keepalive with, why would we want to make them add such regi=
stration logic?  What would be the gain over just simply sending OPTIONS at=
 that point?
>>
>> The beauty of a STUN or double-CRLF keepalive is, in my mind: it's trivi=
al to construct, trivial to parse, very small and fixed size, can be separa=
ted or handled at a lower layer, explicit in its use, and does not get stop=
ped by a SIP-layer overload control.  In short: it's a transport connection=
-layer keepalive, no more no less.  And indicating it in the Via keeps it a=
 hop-link thing, backwards-compatible, little or no provisioning, and with =
no URI target addressing issues.
>>
>> -hadriel
>>
>>   =

>>     =

>>> -----Original Message-----
>>> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
>>> Sent: Wednesday, June 25, 2008 5:56 PM
>>> To: Christer Holmberg
>>> Cc: Hadriel Kaplan; Markus.Isomaki@nokia.com; =

>>> john.elwell@siemens.com; drage@alcatel-lucent.com; sip@ietf.org
>>> Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy =

>>> use case
>>>
>>> No, not a B2BUA. Each element would have a virtual UA function "in =

>>> parallel" to its regular function (e.g. being a proxy). And indeed, =

>>> one way to implement this would be to keep either one (one way) or =

>>> two (opposite ways) flows active between them.
>>>
>>> Like so:
>>>
>>> |-----------|                     |-----------|
>>> |   Proxy     |                     |  Proxy     |
>>> |-----------|                     |-----------|
>>> |     UA      | <---------->|      UA     |
>>> |-----------|                     |-----------|
>>>
>>> Each "UA" would implement both a simple registrar and a UAC =

>>> performing registration. It could also be setup asymmetrically, with =

>>> the (smaller) IP-PBX doing a single registration towards the (bigger) =

>>> Service provider network. Registration expiry would denote a loss of co=
nnectivity.
>>>
>>> Regards,
>>> Jeroen
>>>
>>>
>>> Christer Holmberg wrote:
>>>     =

>>>       =

>>>> Hi,
>>>>
>>>> So, you are proposing that each element should be a B2BUA, and both
>>>>       =

>>>>         =

>>> elements then register towards each other and use Outbound???
>>>     =

>>>       =

>>>> Regards,
>>>>
>>>> Christer
>>>>
>>>> -----Original Message-----
>>>> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
>>>> Sent: 25. kes=E4kuuta 2008 22:57
>>>> To: Hadriel Kaplan
>>>> Cc: Markus.Isomaki@nokia.com; john.elwell@siemens.com; =

>>>> drage@alcatel-
>>>>       =

>>>>         =

>>> lucent.com; sip@ietf.org; Christer Holmberg
>>>     =

>>>       =

>>>> Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy =

>>>> use
>>>>       =

>>>>         =

>>> case
>>>     =

>>>       =

>>>> Hadriel, Markus,
>>>>
>>>> Instead of standardizing keep-alives between proxies, how about we
>>>>       =

>>>>         =

>>> define a "virtual UA" on each element (similar to the one described =

>>> in
>>>     =

>>>       =

>>>> RFC3261 section 16.7 point 6) to be used to provide this functionality?
>>>> (using existing outbound functionality, perhaps both ways)
>>>>
>>>> Regards,
>>>> Jeroen
>>>>
>>>> Hadriel Kaplan wrote:
>>>>
>>>>       =

>>>>         =

>>>>> Yes I am of that same opinion - that any real "IP-PBX" or whatever =

>>>>> big enough NOT to be doing Registration, and to instead do static
>>>>>         =

>>>>>           =

>>> provisioning or DNS, would be given a static hole/DMZ address in =

>>> their firewall/NAT.  But some of my customers have told me otherwise.
>>> (interestingly mostly in APAC region) There's also some concern that =

>>> while a static entry is there for inbound TCP connections, the PBX =

>>> creates outbound ones to the service provider which are ephemeral =

>>> port sources and need to live for very long durations (though why =

>>> they can't just do TCP keepalive is beyond me, but I'm no expert).
>>>     =

>>>       =

>>>>> But anyway, the big issue we've seen is that we need both the PBX =

>>>>> and
>>>>>         =

>>>>>           =

>>> the service provider box to detect failure before an active =

>>> call/request attempt is made; to trigger alternate route selection =

>>> without waiting for transport failure, and as a method to detect livene=
ss again and revert.
>>> Today that's almost exclusively done with Options requests as far as =

>>> I've seen, and lots of people don't seem to like that.
>>>     =

>>>       =

>>>>> -hadriel
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>         =

>>>>>           =

>>>>>> -----Original Message-----
>>>>>> From: Markus.Isomaki@nokia.com [mailto:Markus.Isomaki@nokia.com]
>>>>>> Sent: Wednesday, June 25, 2008 3:11 PM
>>>>>> To: john.elwell@siemens.com; Hadriel Kaplan; =

>>>>>> drage@alcatel-lucent.com; sip@ietf.org
>>>>>> Cc: christer.holmberg@ericsson.com
>>>>>> Subject: RE: [Sip] Progress draft-holmberg-sip-keep: =

>>>>>> proxy-to-proxy use case
>>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> I'm a bit sceptical about the need for keep-alives between proxies.
>>>>>> It is of course entirely possible that an enterprise PBX is =

>>>>>> connected to (or peering with) a service provider proxy through a =

>>>>>> NAT and/or a firewall. However, wouldn't such a NAT or firewall be =

>>>>>> under the administration of either the enterprise itself or its =

>>>>>> ISP (who quite often would be the SIP service provider), and the =

>>>>>> required port forwardings or firewall rules could be set through adm=
inistration.
>>>>>> This means that there would not be need for keepalive traffic to =

>>>>>> implicitely keep the mapping/pinhole open.
>>>>>>
>>>>>> Or are there really deployment cases where there are SIP PBXs =

>>>>>> behind unadministrated NATs or firewalls?
>>>>>>
>>>>>> Wouldn't we then need keepalives for SMTP as well, or how has the =

>>>>>> e-mail infrastructure managed to solve this problem?
>>>>>>
>>>>>> Markus
>>>>>>
>>>>>>
>>>>>>
>>>>>>           =

>>>>>>             =

>>>>> _______________________________________________
>>>>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>>>>> This list is for NEW development of the core SIP Protocol Use =

>>>>> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

>>>>>           =

>>>>       =

>>>>         =

>>   =

>>     =

>
>   =

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


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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
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Hi, =


>The only solution I can think of that would give less processing, would be=
 to simply open a TCP connection (or UDP flow =

>with STUN) without sending any request. Less "overkill", but it would open=
 up a bunch of security issues ( e.g. TCP =

>connection resources being consumed by unauthorized elements, leading to D=
oS; all the concerns with STUNning before =

>checking support, etc. ). So I don't see how sending at least one SIP requ=
est could be avoided. REGISTER is as good as =

>any.
>
>Which proxies to register with: that would of course have to be provisione=
d somehow, this is no different from having to =

>provision which outbound proxy to use

There is a big difference. A UA will always send the requests via the outbo=
und proxy, but a proxy can communicate with an unlimited number of other pr=
oxies. Those proxies are normally found (unless a service route exists) usi=
ng DNS when the proxy receives a request to forward.

>UEs that don't register: the idea is to make all such UEs register, so the=
re'd be no such case left

As said earlier, at least 3GPP has a requirement for non-register emergency=
 calls.

Regards,

Christer





Christer Holmberg wrote:
> Jeroen,
>
> I don't understand what benefit it would be to establish outbound registr=
ations between all SIP entities that want to use the keep-alive/heartbeat. =
Yes, we would "re-use an existing mechanism", but in my opinion it would re=
ally be an overkill,a hack - and anything but keep-it-simple...
>
> Also, how would the proxies know which proxies it should establish regist=
rations which?
>
> Also, it wouldn't work in the use-case with UEs that don't register.
>
> Regards,
>
> Christer
>
>
> -----Original Message-----
> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
> Sent: 26. kes=E4kuuta 2008 9:50
> To: Hadriel Kaplan
> Cc: Christer Holmberg; Markus.Isomaki@nokia.com; =

> john.elwell@siemens.com; drage@alcatel-lucent.com; sip@ietf.org
> Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy =

> use case
>
> Hadriel,
>
> I did not intend to suggest to use the registration interval timer for ke=
ep-alive, of course all the regular flow keep-alive mechanisms of outbound =
could be used.
>
> The benefit is that we would reuse the efforts put in outbound, instead o=
f inventing yet another, only marginally different, mechanism for each spec=
ific case. From the sending element point of view, sending a REGISTER isn't=
 very different from sending an OPTIONS; subsequent keep-alives can then be=
come STUN or double CRLF, until the registration needs to be refreshed.
>
> Using REGISTER instead of OPTIONS also provides the means to use the =

> standard authentication mechanisms, besides reusing outbound (and =

> there's a few other reusable things, like Path headers, but let's keep =

> it simple for the moment)
>
> Regards,
> Jeroen
>
> Hadriel Kaplan wrote:
>   =

>> Hey Jeroen,
>> I'm not sure I follow you.  The SIP Forum SIP-Connect profile, and now T=
ISPAN, make use of a REGISTER for an IP-PBX to register a whole set of AoR'=
s to be routed to.  For those devices that need that ability, it makes sens=
e to use outbound of course, and its inherent keepalive, regardless of bein=
g behind a NAT or not.  And I don't think Christer's suggesting otherwise.
>>
>> But for IP-PBX's or proxies which do not themselves represent an AoR tar=
get of requests, or do not represent one in the domain of the device they w=
ant to perform keepalive with, why would we want to make them add such regi=
stration logic?  What would be the gain over just simply sending OPTIONS at=
 that point?
>>
>> The beauty of a STUN or double-CRLF keepalive is, in my mind: it's trivi=
al to construct, trivial to parse, very small and fixed size, can be separa=
ted or handled at a lower layer, explicit in its use, and does not get stop=
ped by a SIP-layer overload control.  In short: it's a transport connection=
-layer keepalive, no more no less.  And indicating it in the Via keeps it a=
 hop-link thing, backwards-compatible, little or no provisioning, and with =
no URI target addressing issues.
>>
>> -hadriel
>>
>>   =

>>     =

>>> -----Original Message-----
>>> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
>>> Sent: Wednesday, June 25, 2008 5:56 PM
>>> To: Christer Holmberg
>>> Cc: Hadriel Kaplan; Markus.Isomaki@nokia.com; =

>>> john.elwell@siemens.com; drage@alcatel-lucent.com; sip@ietf.org
>>> Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy =

>>> use case
>>>
>>> No, not a B2BUA. Each element would have a virtual UA function "in =

>>> parallel" to its regular function (e.g. being a proxy). And indeed, =

>>> one way to implement this would be to keep either one (one way) or =

>>> two (opposite ways) flows active between them.
>>>
>>> Like so:
>>>
>>> |-----------|                     |-----------|
>>> |   Proxy     |                     |  Proxy     |
>>> |-----------|                     |-----------|
>>> |     UA      | <---------->|      UA     |
>>> |-----------|                     |-----------|
>>>
>>> Each "UA" would implement both a simple registrar and a UAC =

>>> performing registration. It could also be setup asymmetrically, with =

>>> the (smaller) IP-PBX doing a single registration towards the =

>>> (bigger) Service provider network. Registration expiry would denote a l=
oss of connectivity.
>>>
>>> Regards,
>>> Jeroen
>>>
>>>
>>> Christer Holmberg wrote:
>>>     =

>>>       =

>>>> Hi,
>>>>
>>>> So, you are proposing that each element should be a B2BUA, and both
>>>>       =

>>>>         =

>>> elements then register towards each other and use Outbound???
>>>     =

>>>       =

>>>> Regards,
>>>>
>>>> Christer
>>>>
>>>> -----Original Message-----
>>>> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
>>>> Sent: 25. kes=E4kuuta 2008 22:57
>>>> To: Hadriel Kaplan
>>>> Cc: Markus.Isomaki@nokia.com; john.elwell@siemens.com;
>>>> drage@alcatel-
>>>>       =

>>>>         =

>>> lucent.com; sip@ietf.org; Christer Holmberg
>>>     =

>>>       =

>>>> Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy =

>>>> use
>>>>       =

>>>>         =

>>> case
>>>     =

>>>       =

>>>> Hadriel, Markus,
>>>>
>>>> Instead of standardizing keep-alives between proxies, how about we
>>>>       =

>>>>         =

>>> define a "virtual UA" on each element (similar to the one described =

>>> in
>>>     =

>>>       =

>>>> RFC3261 section 16.7 point 6) to be used to provide this functionality?
>>>> (using existing outbound functionality, perhaps both ways)
>>>>
>>>> Regards,
>>>> Jeroen
>>>>
>>>> Hadriel Kaplan wrote:
>>>>
>>>>       =

>>>>         =

>>>>> Yes I am of that same opinion - that any real "IP-PBX" or whatever =

>>>>> big enough NOT to be doing Registration, and to instead do static
>>>>>         =

>>>>>           =

>>> provisioning or DNS, would be given a static hole/DMZ address in =

>>> their firewall/NAT.  But some of my customers have told me otherwise.
>>> (interestingly mostly in APAC region) There's also some concern that =

>>> while a static entry is there for inbound TCP connections, the PBX =

>>> creates outbound ones to the service provider which are ephemeral =

>>> port sources and need to live for very long durations (though why =

>>> they can't just do TCP keepalive is beyond me, but I'm no expert).
>>>     =

>>>       =

>>>>> But anyway, the big issue we've seen is that we need both the PBX =

>>>>> and
>>>>>         =

>>>>>           =

>>> the service provider box to detect failure before an active =

>>> call/request attempt is made; to trigger alternate route selection =

>>> without waiting for transport failure, and as a method to detect livene=
ss again and revert.
>>> Today that's almost exclusively done with Options requests as far as =

>>> I've seen, and lots of people don't seem to like that.
>>>     =

>>>       =

>>>>> -hadriel
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>         =

>>>>>           =

>>>>>> -----Original Message-----
>>>>>> From: Markus.Isomaki@nokia.com [mailto:Markus.Isomaki@nokia.com]
>>>>>> Sent: Wednesday, June 25, 2008 3:11 PM
>>>>>> To: john.elwell@siemens.com; Hadriel Kaplan; =

>>>>>> drage@alcatel-lucent.com; sip@ietf.org
>>>>>> Cc: christer.holmberg@ericsson.com
>>>>>> Subject: RE: [Sip] Progress draft-holmberg-sip-keep: =

>>>>>> proxy-to-proxy use case
>>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> I'm a bit sceptical about the need for keep-alives between proxies.
>>>>>> It is of course entirely possible that an enterprise PBX is =

>>>>>> connected to (or peering with) a service provider proxy through a =

>>>>>> NAT and/or a firewall. However, wouldn't such a NAT or firewall =

>>>>>> be under the administration of either the enterprise itself or =

>>>>>> its ISP (who quite often would be the SIP service provider), and =

>>>>>> the required port forwardings or firewall rules could be set through=
 administration.
>>>>>> This means that there would not be need for keepalive traffic to =

>>>>>> implicitely keep the mapping/pinhole open.
>>>>>>
>>>>>> Or are there really deployment cases where there are SIP PBXs =

>>>>>> behind unadministrated NATs or firewalls?
>>>>>>
>>>>>> Wouldn't we then need keepalives for SMTP as well, or how has the =

>>>>>> e-mail infrastructure managed to solve this problem?
>>>>>>
>>>>>> Markus
>>>>>>
>>>>>>
>>>>>>
>>>>>>           =

>>>>>>             =

>>>>> _______________________________________________
>>>>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>>>>> This list is for NEW development of the core SIP Protocol Use =

>>>>> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

>>>>>           =

>>>>       =

>>>>         =

>>   =

>>     =

>
>   =

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


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From: Jeroen van Bemmel <jbemmel@zonnet.nl>
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Cc: sip@ietf.org, Markus.Isomaki@nokia.com, drage@alcatel-lucent.com,
	Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use case
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John,

The problem with "not appropriate" is that it's subjective and =

ambiguous. I think we understand "not strictly needed" (or "overkill" as =

some say), e.g. when there is no NAT device in between.

A case I can imagine where outbound wouldn't work, is if the proxies in =

question are connected indirectly (i.e. not hop-by-hop). Then the =

outbound keep-alive mechanism doesn't work, and you'd need some SIP =

request instead (or some elaborate scheme to pass on keep-alive =

information across multiple hops in some other way)

Regards,
Jeroen

Elwell, John wrote:
> Jeroen,
>
> Yes, this is effectively what you have when an IPPBX is required to regis=
ter with a service provider. But in that case, there might also be a need t=
o implement sip-outbound if there are unadministered NATs/firewalls in the =
way. Indeed, sip-outbound would operate between a virtual UA on the IPPBX s=
ide and a registrar on the service provider side.
>
> What I am interested in gaining from this discussion is whether there are=
 proxy-to-proxy cases where sip-outbound is not appropriate but keep-alive =
(or heartbeat) is needed.
>
> John
>
>   =

>> -----Original Message-----
>> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl] =

>> Sent: 25 June 2008 22:56
>> To: Christer Holmberg
>> Cc: Hadriel Kaplan; Markus.Isomaki@nokia.com; Elwell, John; =

>> drage@alcatel-lucent.com; sip@ietf.org
>> Subject: Re: [Sip] Progress draft-holmberg-sip-keep: =

>> proxy-to-proxy use case
>>
>> No, not a B2BUA. Each element would have a virtual UA function "in =

>> parallel" to its regular function (e.g. being a proxy). And =

>> indeed, one =

>> way to implement this would be to keep either one (one way) or two =

>> (opposite ways) flows active between them.
>>
>> Like so:
>>
>> |-----------|                     |-----------|
>> |   Proxy     |                     |  Proxy     |
>> |-----------|                     |-----------|
>> |     UA      | <---------->|      UA     |
>> |-----------|                     |-----------|
>>
>> Each "UA" would implement both a simple registrar and a UAC =

>> performing =

>> registration. It could also be setup asymmetrically, with the =

>> (smaller) =

>> IP-PBX doing a single registration towards the (bigger) =

>> Service provider =

>> network. Registration expiry would denote a loss of connectivity.
>>
>> Regards,
>> Jeroen
>>
>>
>> Christer Holmberg wrote:
>>     =

>>> Hi,
>>>
>>> So, you are proposing that each element should be a B2BUA, =

>>>       =

>> and both elements then register towards each other and use Outbound???
>>     =

>>> Regards,
>>>
>>> Christer =

>>>
>>> -----Original Message-----
>>> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl] =

>>> Sent: 25. kes=E4kuuta 2008 22:57
>>> To: Hadriel Kaplan
>>> Cc: Markus.Isomaki@nokia.com; john.elwell@siemens.com; =

>>>       =

>> drage@alcatel-lucent.com; sip@ietf.org; Christer Holmberg
>>     =

>>> Subject: Re: [Sip] Progress draft-holmberg-sip-keep: =

>>>       =

>> proxy-to-proxy use case
>>     =

>>> Hadriel, Markus,
>>>
>>> Instead of standardizing keep-alives between proxies, how =

>>>       =

>> about we define a "virtual UA" on each element (similar to =

>> the one described in
>>     =

>>> RFC3261 section 16.7 point 6) to be used to provide this =

>>>       =

>> functionality? =

>>     =

>>> (using existing outbound functionality, perhaps both ways)
>>>
>>> Regards,
>>> Jeroen
>>>
>>> Hadriel Kaplan wrote:
>>>   =

>>>       =

>>>> Yes I am of that same opinion - that any real "IP-PBX" or =

>>>>         =

>> whatever big =

>>     =

>>>> enough NOT to be doing Registration, and to instead do =

>>>>         =

>> static provisioning or DNS, would be given a static hole/DMZ =

>> address in their firewall/NAT.  But some of my customers have =

>> told me otherwise. (interestingly mostly in APAC region) =

>> There's also some concern that while a static entry is there =

>> for inbound TCP connections, the PBX creates outbound ones to =

>> the service provider which are ephemeral port sources and =

>> need to live for very long durations (though why they can't =

>> just do TCP keepalive is beyond me, but I'm no expert).
>>     =

>>>> But anyway, the big issue we've seen is that we need both =

>>>>         =

>> the PBX and the service provider box to detect failure before =

>> an active call/request attempt is made; to trigger alternate =

>> route selection without waiting for transport failure, and as =

>> a method to detect liveness again and revert.  Today that's =

>> almost exclusively done with Options requests as far as I've =

>> seen, and lots of people don't seem to like that.
>>     =

>>>> -hadriel
>>>>
>>>>
>>>>   =

>>>>     =

>>>>         =

>>>>> -----Original Message-----
>>>>> From: Markus.Isomaki@nokia.com [mailto:Markus.Isomaki@nokia.com]
>>>>> Sent: Wednesday, June 25, 2008 3:11 PM
>>>>> To: john.elwell@siemens.com; Hadriel Kaplan; =

>>>>> drage@alcatel-lucent.com; sip@ietf.org
>>>>> Cc: christer.holmberg@ericsson.com
>>>>> Subject: RE: [Sip] Progress draft-holmberg-sip-keep: =

>>>>>           =

>> proxy-to-proxy =

>>     =

>>>>> use case
>>>>>
>>>>> Hi,
>>>>>
>>>>> I'm a bit sceptical about the need for keep-alives =

>>>>>           =

>> between proxies. =

>>     =

>>>>> It is of course entirely possible that an enterprise PBX =

>>>>>           =

>> is connected =

>>     =

>>>>> to (or peering with) a service provider proxy through a =

>>>>>           =

>> NAT and/or a =

>>     =

>>>>> firewall. However, wouldn't such a NAT or firewall be under the =

>>>>> administration of either the enterprise itself or its ISP =

>>>>>           =

>> (who quite =

>>     =

>>>>> often would be the SIP service provider), and the required port =

>>>>> forwardings or firewall rules could be set through =

>>>>>           =

>> administration. =

>>     =

>>>>> This means that there would not be need for keepalive traffic to =

>>>>> implicitely keep the mapping/pinhole open.
>>>>>
>>>>> Or are there really deployment cases where there are SIP =

>>>>>           =

>> PBXs behind =

>>     =

>>>>> unadministrated NATs or firewalls?
>>>>>
>>>>> Wouldn't we then need keepalives for SMTP as well, or how has the =

>>>>> e-mail infrastructure managed to solve this problem?
>>>>>
>>>>> Markus
>>>>>
>>>>>     =

>>>>>       =

>>>>>           =

>>>> _______________________________________________
>>>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>>>> This list is for NEW development of the core SIP Protocol Use =

>>>> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

>>>>     =

>>>>         =

>>>   =

>>>       =

>
>   =

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


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From: Jeroen van Bemmel <jbemmel@zonnet.nl>
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: sip@ietf.org, drage@alcatel-lucent.com, john.elwell@siemens.com,
	Markus.Isomaki@nokia.com
Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use case
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Christer,

We're mixing different cases here. The 3GPP non-register emergency call =

scenario is for an endpoint, I was talking about the proxy-to-proxy case =

(I believe the 3GPP case can be addressed using standard mechanisms =

available today, it's only about authorization policies and =

identification of what calls are emergency calls).

It doesn't make sense for a proxy to keep-alive connections with =

arbitrary other proxies. Either it has a limited set of peers (such as =

in case of an IP-PBX peering with a service provider), or it routes =

using DNS based on the domain in the request URI (or top Route header). =

For the latter case, the act of sending the INVITE itself will detect =

reachability.

So I suggest to limit the proxy-to-proxy use-case for keep-alive to =

proxies that route towards a limited set of next hops (instead of DNS on =

domain). According to RFC3261, such proxies would insert a Route header =

(with loose routing)

Regards,
Jeroen

Christer Holmberg wrote:
> Hi, =

>
>   =

>> The only solution I can think of that would give less processing, would =
be to simply open a TCP connection (or UDP flow =

>> with STUN) without sending any request. Less "overkill", but it would op=
en up a bunch of security issues ( e.g. TCP =

>> connection resources being consumed by unauthorized elements, leading to=
 DoS; all the concerns with STUNning before =

>> checking support, etc. ). So I don't see how sending at least one SIP re=
quest could be avoided. REGISTER is as good as =

>> any.
>>
>> Which proxies to register with: that would of course have to be provisio=
ned somehow, this is no different from having to =

>> provision which outbound proxy to use
>>     =

>
> There is a big difference. A UA will always send the requests via the out=
bound proxy, but a proxy can communicate with an unlimited number of other =
proxies. Those proxies are normally found (unless a service route exists) u=
sing DNS when the proxy receives a request to forward.
>
>   =

>> UEs that don't register: the idea is to make all such UEs register, so t=
here'd be no such case left
>>     =

>
> As said earlier, at least 3GPP has a requirement for non-register emergen=
cy calls.
>
> Regards,
>
> Christer
>
>
>
>
>
> Christer Holmberg wrote:
>   =

>> Jeroen,
>>
>> I don't understand what benefit it would be to establish outbound regist=
rations between all SIP entities that want to use the keep-alive/heartbeat.=
 Yes, we would "re-use an existing mechanism", but in my opinion it would r=
eally be an overkill,a hack - and anything but keep-it-simple...
>>
>> Also, how would the proxies know which proxies it should establish regis=
trations which?
>>
>> Also, it wouldn't work in the use-case with UEs that don't register.
>>
>> Regards,
>>
>> Christer
>>
>>
>> -----Original Message-----
>> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
>> Sent: 26. kes=E4kuuta 2008 9:50
>> To: Hadriel Kaplan
>> Cc: Christer Holmberg; Markus.Isomaki@nokia.com; =

>> john.elwell@siemens.com; drage@alcatel-lucent.com; sip@ietf.org
>> Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy =

>> use case
>>
>> Hadriel,
>>
>> I did not intend to suggest to use the registration interval timer for k=
eep-alive, of course all the regular flow keep-alive mechanisms of outbound=
 could be used.
>>
>> The benefit is that we would reuse the efforts put in outbound, instead =
of inventing yet another, only marginally different, mechanism for each spe=
cific case. From the sending element point of view, sending a REGISTER isn'=
t very different from sending an OPTIONS; subsequent keep-alives can then b=
ecome STUN or double CRLF, until the registration needs to be refreshed.
>>
>> Using REGISTER instead of OPTIONS also provides the means to use the =

>> standard authentication mechanisms, besides reusing outbound (and =

>> there's a few other reusable things, like Path headers, but let's keep =

>> it simple for the moment)
>>
>> Regards,
>> Jeroen
>>
>> Hadriel Kaplan wrote:
>>   =

>>     =

>>> Hey Jeroen,
>>> I'm not sure I follow you.  The SIP Forum SIP-Connect profile, and now =
TISPAN, make use of a REGISTER for an IP-PBX to register a whole set of AoR=
's to be routed to.  For those devices that need that ability, it makes sen=
se to use outbound of course, and its inherent keepalive, regardless of bei=
ng behind a NAT or not.  And I don't think Christer's suggesting otherwise.
>>>
>>> But for IP-PBX's or proxies which do not themselves represent an AoR ta=
rget of requests, or do not represent one in the domain of the device they =
want to perform keepalive with, why would we want to make them add such reg=
istration logic?  What would be the gain over just simply sending OPTIONS a=
t that point?
>>>
>>> The beauty of a STUN or double-CRLF keepalive is, in my mind: it's triv=
ial to construct, trivial to parse, very small and fixed size, can be separ=
ated or handled at a lower layer, explicit in its use, and does not get sto=
pped by a SIP-layer overload control.  In short: it's a transport connectio=
n-layer keepalive, no more no less.  And indicating it in the Via keeps it =
a hop-link thing, backwards-compatible, little or no provisioning, and with=
 no URI target addressing issues.
>>>
>>> -hadriel
>>>
>>>   =

>>>     =

>>>       =

>>>> -----Original Message-----
>>>> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
>>>> Sent: Wednesday, June 25, 2008 5:56 PM
>>>> To: Christer Holmberg
>>>> Cc: Hadriel Kaplan; Markus.Isomaki@nokia.com; =

>>>> john.elwell@siemens.com; drage@alcatel-lucent.com; sip@ietf.org
>>>> Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy =

>>>> use case
>>>>
>>>> No, not a B2BUA. Each element would have a virtual UA function "in =

>>>> parallel" to its regular function (e.g. being a proxy). And indeed, =

>>>> one way to implement this would be to keep either one (one way) or =

>>>> two (opposite ways) flows active between them.
>>>>
>>>> Like so:
>>>>
>>>> |-----------|                     |-----------|
>>>> |   Proxy     |                     |  Proxy     |
>>>> |-----------|                     |-----------|
>>>> |     UA      | <---------->|      UA     |
>>>> |-----------|                     |-----------|
>>>>
>>>> Each "UA" would implement both a simple registrar and a UAC =

>>>> performing registration. It could also be setup asymmetrically, with =

>>>> the (smaller) IP-PBX doing a single registration towards the =

>>>> (bigger) Service provider network. Registration expiry would denote a =
loss of connectivity.
>>>>
>>>> Regards,
>>>> Jeroen
>>>>
>>>>
>>>> Christer Holmberg wrote:
>>>>     =

>>>>       =

>>>>         =

>>>>> Hi,
>>>>>
>>>>> So, you are proposing that each element should be a B2BUA, and both
>>>>>       =

>>>>>         =

>>>>>           =

>>>> elements then register towards each other and use Outbound???
>>>>     =

>>>>       =

>>>>         =

>>>>> Regards,
>>>>>
>>>>> Christer
>>>>>
>>>>> -----Original Message-----
>>>>> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
>>>>> Sent: 25. kes=E4kuuta 2008 22:57
>>>>> To: Hadriel Kaplan
>>>>> Cc: Markus.Isomaki@nokia.com; john.elwell@siemens.com;
>>>>> drage@alcatel-
>>>>>       =

>>>>>         =

>>>>>           =

>>>> lucent.com; sip@ietf.org; Christer Holmberg
>>>>     =

>>>>       =

>>>>         =

>>>>> Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy =

>>>>> use
>>>>>       =

>>>>>         =

>>>>>           =

>>>> case
>>>>     =

>>>>       =

>>>>         =

>>>>> Hadriel, Markus,
>>>>>
>>>>> Instead of standardizing keep-alives between proxies, how about we
>>>>>       =

>>>>>         =

>>>>>           =

>>>> define a "virtual UA" on each element (similar to the one described =

>>>> in
>>>>     =

>>>>       =

>>>>         =

>>>>> RFC3261 section 16.7 point 6) to be used to provide this functionalit=
y?
>>>>> (using existing outbound functionality, perhaps both ways)
>>>>>
>>>>> Regards,
>>>>> Jeroen
>>>>>
>>>>> Hadriel Kaplan wrote:
>>>>>
>>>>>       =

>>>>>         =

>>>>>           =

>>>>>> Yes I am of that same opinion - that any real "IP-PBX" or whatever =

>>>>>> big enough NOT to be doing Registration, and to instead do static
>>>>>>         =

>>>>>>           =

>>>>>>             =

>>>> provisioning or DNS, would be given a static hole/DMZ address in =

>>>> their firewall/NAT.  But some of my customers have told me otherwise.
>>>> (interestingly mostly in APAC region) There's also some concern that =

>>>> while a static entry is there for inbound TCP connections, the PBX =

>>>> creates outbound ones to the service provider which are ephemeral =

>>>> port sources and need to live for very long durations (though why =

>>>> they can't just do TCP keepalive is beyond me, but I'm no expert).
>>>>     =

>>>>       =

>>>>         =

>>>>>> But anyway, the big issue we've seen is that we need both the PBX =

>>>>>> and
>>>>>>         =

>>>>>>           =

>>>>>>             =

>>>> the service provider box to detect failure before an active =

>>>> call/request attempt is made; to trigger alternate route selection =

>>>> without waiting for transport failure, and as a method to detect liven=
ess again and revert.
>>>> Today that's almost exclusively done with Options requests as far as =

>>>> I've seen, and lots of people don't seem to like that.
>>>>     =

>>>>       =

>>>>         =

>>>>>> -hadriel
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>         =

>>>>>>           =

>>>>>>             =

>>>>>>> -----Original Message-----
>>>>>>> From: Markus.Isomaki@nokia.com [mailto:Markus.Isomaki@nokia.com]
>>>>>>> Sent: Wednesday, June 25, 2008 3:11 PM
>>>>>>> To: john.elwell@siemens.com; Hadriel Kaplan; =

>>>>>>> drage@alcatel-lucent.com; sip@ietf.org
>>>>>>> Cc: christer.holmberg@ericsson.com
>>>>>>> Subject: RE: [Sip] Progress draft-holmberg-sip-keep: =

>>>>>>> proxy-to-proxy use case
>>>>>>>
>>>>>>> Hi,
>>>>>>>
>>>>>>> I'm a bit sceptical about the need for keep-alives between proxies.
>>>>>>> It is of course entirely possible that an enterprise PBX is =

>>>>>>> connected to (or peering with) a service provider proxy through a =

>>>>>>> NAT and/or a firewall. However, wouldn't such a NAT or firewall =

>>>>>>> be under the administration of either the enterprise itself or =

>>>>>>> its ISP (who quite often would be the SIP service provider), and =

>>>>>>> the required port forwardings or firewall rules could be set throug=
h administration.
>>>>>>> This means that there would not be need for keepalive traffic to =

>>>>>>> implicitely keep the mapping/pinhole open.
>>>>>>>
>>>>>>> Or are there really deployment cases where there are SIP PBXs =

>>>>>>> behind unadministrated NATs or firewalls?
>>>>>>>
>>>>>>> Wouldn't we then need keepalives for SMTP as well, or how has the =

>>>>>>> e-mail infrastructure managed to solve this problem?
>>>>>>>
>>>>>>> Markus
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>           =

>>>>>>>             =

>>>>>>>               =

>>>>>> _______________________________________________
>>>>>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>>>>>> This list is for NEW development of the core SIP Protocol Use =

>>>>>> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

>>>>>>           =

>>>>>>             =

>>>>>       =

>>>>>         =

>>>>>           =

>>>   =

>>>     =

>>>       =

>>   =

>>     =

>
>   =

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


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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
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	Markus.Isomaki@nokia.com
Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use case
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Hi, =


>We're mixing different cases here. The 3GPP non-register emergency call sc=
enario is for an endpoint, I was talking about =

>the proxy-to-proxy case (I believe the 3GPP case can be addressed using st=
andard mechanisms available today, it's only =

>about authorization policies and identification of what calls are emergenc=
y calls).
>
>It doesn't make sense for a proxy to keep-alive connections with arbitrary=
 other proxies. Either it has a limited set of =

>peers (such as in case of an IP-PBX peering with a service provider), or i=
t routes using DNS based on the domain in the =

>request URI (or top Route header). =

>For the latter case, the act of sending the INVITE itself will detect reac=
hability.
>
>So I suggest to limit the proxy-to-proxy use-case for keep-alive to proxie=
s that route towards a limited set of next hops =

>(instead of DNS on domain). According to RFC3261, such proxies would inser=
t a Route header (with loose routing)

Ok. Thanks for the clarification.

So, if I understand you correctly you either:

1. Propose that proxy-to-proxy should not be part of the scope of the draft
2. Proxy-to-proxy could be part of the scope of the draft, but we would not=
 necessarily use the same mechanism for UA-to-proxy and proxy-to-proxy.

Correct?

Regards,

Christer



Christer Holmberg wrote:
> Hi,
>
>   =

>> The only solution I can think of that would give less processing, =

>> would be to simply open a TCP connection (or UDP flow with STUN) =

>> without sending any request. Less "overkill", but it would open up a =

>> bunch of security issues ( e.g. TCP connection resources being =

>> consumed by unauthorized elements, leading to DoS; all the concerns with=
 STUNning before checking support, etc. ). So I don't see how sending at le=
ast one SIP request could be avoided. REGISTER is as good as any.
>>
>> Which proxies to register with: that would of course have to be =

>> provisioned somehow, this is no different from having to provision =

>> which outbound proxy to use
>>     =

>
> There is a big difference. A UA will always send the requests via the out=
bound proxy, but a proxy can communicate with an unlimited number of other =
proxies. Those proxies are normally found (unless a service route exists) u=
sing DNS when the proxy receives a request to forward.
>
>   =

>> UEs that don't register: the idea is to make all such UEs register, =

>> so there'd be no such case left
>>     =

>
> As said earlier, at least 3GPP has a requirement for non-register emergen=
cy calls.
>
> Regards,
>
> Christer
>
>
>
>
>
> Christer Holmberg wrote:
>   =

>> Jeroen,
>>
>> I don't understand what benefit it would be to establish outbound regist=
rations between all SIP entities that want to use the keep-alive/heartbeat.=
 Yes, we would "re-use an existing mechanism", but in my opinion it would r=
eally be an overkill,a hack - and anything but keep-it-simple...
>>
>> Also, how would the proxies know which proxies it should establish regis=
trations which?
>>
>> Also, it wouldn't work in the use-case with UEs that don't register.
>>
>> Regards,
>>
>> Christer
>>
>>
>> -----Original Message-----
>> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
>> Sent: 26. kes=E4kuuta 2008 9:50
>> To: Hadriel Kaplan
>> Cc: Christer Holmberg; Markus.Isomaki@nokia.com; =

>> john.elwell@siemens.com; drage@alcatel-lucent.com; sip@ietf.org
>> Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy =

>> use case
>>
>> Hadriel,
>>
>> I did not intend to suggest to use the registration interval timer for k=
eep-alive, of course all the regular flow keep-alive mechanisms of outbound=
 could be used.
>>
>> The benefit is that we would reuse the efforts put in outbound, instead =
of inventing yet another, only marginally different, mechanism for each spe=
cific case. From the sending element point of view, sending a REGISTER isn'=
t very different from sending an OPTIONS; subsequent keep-alives can then b=
ecome STUN or double CRLF, until the registration needs to be refreshed.
>>
>> Using REGISTER instead of OPTIONS also provides the means to use the =

>> standard authentication mechanisms, besides reusing outbound (and =

>> there's a few other reusable things, like Path headers, but let's =

>> keep it simple for the moment)
>>
>> Regards,
>> Jeroen
>>
>> Hadriel Kaplan wrote:
>>   =

>>     =

>>> Hey Jeroen,
>>> I'm not sure I follow you.  The SIP Forum SIP-Connect profile, and now =
TISPAN, make use of a REGISTER for an IP-PBX to register a whole set of AoR=
's to be routed to.  For those devices that need that ability, it makes sen=
se to use outbound of course, and its inherent keepalive, regardless of bei=
ng behind a NAT or not.  And I don't think Christer's suggesting otherwise.
>>>
>>> But for IP-PBX's or proxies which do not themselves represent an AoR ta=
rget of requests, or do not represent one in the domain of the device they =
want to perform keepalive with, why would we want to make them add such reg=
istration logic?  What would be the gain over just simply sending OPTIONS a=
t that point?
>>>
>>> The beauty of a STUN or double-CRLF keepalive is, in my mind: it's triv=
ial to construct, trivial to parse, very small and fixed size, can be separ=
ated or handled at a lower layer, explicit in its use, and does not get sto=
pped by a SIP-layer overload control.  In short: it's a transport connectio=
n-layer keepalive, no more no less.  And indicating it in the Via keeps it =
a hop-link thing, backwards-compatible, little or no provisioning, and with=
 no URI target addressing issues.
>>>
>>> -hadriel
>>>
>>>   =

>>>     =

>>>       =

>>>> -----Original Message-----
>>>> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
>>>> Sent: Wednesday, June 25, 2008 5:56 PM
>>>> To: Christer Holmberg
>>>> Cc: Hadriel Kaplan; Markus.Isomaki@nokia.com; =

>>>> john.elwell@siemens.com; drage@alcatel-lucent.com; sip@ietf.org
>>>> Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy =

>>>> use case
>>>>
>>>> No, not a B2BUA. Each element would have a virtual UA function "in =

>>>> parallel" to its regular function (e.g. being a proxy). And indeed, =

>>>> one way to implement this would be to keep either one (one way) or =

>>>> two (opposite ways) flows active between them.
>>>>
>>>> Like so:
>>>>
>>>> |-----------|                     |-----------|
>>>> |   Proxy     |                     |  Proxy     |
>>>> |-----------|                     |-----------|
>>>> |     UA      | <---------->|      UA     |
>>>> |-----------|                     |-----------|
>>>>
>>>> Each "UA" would implement both a simple registrar and a UAC =

>>>> performing registration. It could also be setup asymmetrically, =

>>>> with the (smaller) IP-PBX doing a single registration towards the
>>>> (bigger) Service provider network. Registration expiry would denote a =
loss of connectivity.
>>>>
>>>> Regards,
>>>> Jeroen
>>>>
>>>>
>>>> Christer Holmberg wrote:
>>>>     =

>>>>       =

>>>>         =

>>>>> Hi,
>>>>>
>>>>> So, you are proposing that each element should be a B2BUA, and =

>>>>> both
>>>>>       =

>>>>>         =

>>>>>           =

>>>> elements then register towards each other and use Outbound???
>>>>     =

>>>>       =

>>>>         =

>>>>> Regards,
>>>>>
>>>>> Christer
>>>>>
>>>>> -----Original Message-----
>>>>> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
>>>>> Sent: 25. kes=E4kuuta 2008 22:57
>>>>> To: Hadriel Kaplan
>>>>> Cc: Markus.Isomaki@nokia.com; john.elwell@siemens.com;
>>>>> drage@alcatel-
>>>>>       =

>>>>>         =

>>>>>           =

>>>> lucent.com; sip@ietf.org; Christer Holmberg
>>>>     =

>>>>       =

>>>>         =

>>>>> Subject: Re: [Sip] Progress draft-holmberg-sip-keep: =

>>>>> proxy-to-proxy use
>>>>>       =

>>>>>         =

>>>>>           =

>>>> case
>>>>     =

>>>>       =

>>>>         =

>>>>> Hadriel, Markus,
>>>>>
>>>>> Instead of standardizing keep-alives between proxies, how about we
>>>>>       =

>>>>>         =

>>>>>           =

>>>> define a "virtual UA" on each element (similar to the one described =

>>>> in
>>>>     =

>>>>       =

>>>>         =

>>>>> RFC3261 section 16.7 point 6) to be used to provide this functionalit=
y?
>>>>> (using existing outbound functionality, perhaps both ways)
>>>>>
>>>>> Regards,
>>>>> Jeroen
>>>>>
>>>>> Hadriel Kaplan wrote:
>>>>>
>>>>>       =

>>>>>         =

>>>>>           =

>>>>>> Yes I am of that same opinion - that any real "IP-PBX" or =

>>>>>> whatever big enough NOT to be doing Registration, and to instead =

>>>>>> do static
>>>>>>         =

>>>>>>           =

>>>>>>             =

>>>> provisioning or DNS, would be given a static hole/DMZ address in =

>>>> their firewall/NAT.  But some of my customers have told me otherwise.
>>>> (interestingly mostly in APAC region) There's also some concern =

>>>> that while a static entry is there for inbound TCP connections, the =

>>>> PBX creates outbound ones to the service provider which are =

>>>> ephemeral port sources and need to live for very long durations =

>>>> (though why they can't just do TCP keepalive is beyond me, but I'm no =
expert).
>>>>     =

>>>>       =

>>>>         =

>>>>>> But anyway, the big issue we've seen is that we need both the PBX =

>>>>>> and
>>>>>>         =

>>>>>>           =

>>>>>>             =

>>>> the service provider box to detect failure before an active =

>>>> call/request attempt is made; to trigger alternate route selection =

>>>> without waiting for transport failure, and as a method to detect liven=
ess again and revert.
>>>> Today that's almost exclusively done with Options requests as far =

>>>> as I've seen, and lots of people don't seem to like that.
>>>>     =

>>>>       =

>>>>         =

>>>>>> -hadriel
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>         =

>>>>>>           =

>>>>>>             =

>>>>>>> -----Original Message-----
>>>>>>> From: Markus.Isomaki@nokia.com [mailto:Markus.Isomaki@nokia.com]
>>>>>>> Sent: Wednesday, June 25, 2008 3:11 PM
>>>>>>> To: john.elwell@siemens.com; Hadriel Kaplan; =

>>>>>>> drage@alcatel-lucent.com; sip@ietf.org
>>>>>>> Cc: christer.holmberg@ericsson.com
>>>>>>> Subject: RE: [Sip] Progress draft-holmberg-sip-keep: =

>>>>>>> proxy-to-proxy use case
>>>>>>>
>>>>>>> Hi,
>>>>>>>
>>>>>>> I'm a bit sceptical about the need for keep-alives between proxies.
>>>>>>> It is of course entirely possible that an enterprise PBX is =

>>>>>>> connected to (or peering with) a service provider proxy through =

>>>>>>> a NAT and/or a firewall. However, wouldn't such a NAT or =

>>>>>>> firewall be under the administration of either the enterprise =

>>>>>>> itself or its ISP (who quite often would be the SIP service =

>>>>>>> provider), and the required port forwardings or firewall rules coul=
d be set through administration.
>>>>>>> This means that there would not be need for keepalive traffic to =

>>>>>>> implicitely keep the mapping/pinhole open.
>>>>>>>
>>>>>>> Or are there really deployment cases where there are SIP PBXs =

>>>>>>> behind unadministrated NATs or firewalls?
>>>>>>>
>>>>>>> Wouldn't we then need keepalives for SMTP as well, or how has =

>>>>>>> the e-mail infrastructure managed to solve this problem?
>>>>>>>
>>>>>>> Markus
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>           =

>>>>>>>             =

>>>>>>>               =

>>>>>> _______________________________________________
>>>>>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>>>>>> This list is for NEW development of the core SIP Protocol Use =

>>>>>> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

>>>>>>           =

>>>>>>             =

>>>>>       =

>>>>>         =

>>>>>           =

>>>   =

>>>     =

>>>       =

>>   =

>>     =

>
>   =

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


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Yes, each m line start a new media section in the SDP. You can take a
look at SDP RFC.
=20

Best Regards,=20
Ashish Saxena=20
877-5570=20

=20

________________________________

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Sudhir Kumar Reddy
Sent: Thursday, June 26, 2008 12:04 PM
To: sip@ietf.org
Subject: [Sip] can SDP contain 2 'M' lines


Hi All,

In SDP can we have two 'M' lines for re-INVITE i.e. for call hold.

if yes what is RFC / draft

Any response is highly appreciated

Regards
Sudhir=20

------_=_NextPart_001_01C8D763.C0E9EA3B
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<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D908220708-26062008><FONT =
face=3DVerdana=20
color=3D#800000 size=3D2>Yes, each m line start a new media section in =
the SDP. You=20
can take a look at SDP RFC.</FONT></SPAN></DIV>
<DIV>&nbsp;</DIV><!-- Converted from text/rtf format -->
<P><SPAN lang=3Den-us><FONT face=3DVerdana color=3D#800000 size=3D2>Best =

Regards,</FONT></SPAN> <BR><SPAN lang=3Den-us><FONT face=3DVerdana =
color=3D#800000=20
size=3D2>Ashish Saxena</FONT></SPAN> <BR><SPAN lang=3Den-us><FONT =
face=3DVerdana=20
color=3D#800000 size=3D2>877-5570</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> sip-bounces@ietf.org=20
[mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Sudhir Kumar=20
Reddy<BR><B>Sent:</B> Thursday, June 26, 2008 12:04 PM<BR><B>To:</B>=20
sip@ietf.org<BR><B>Subject:</B> [Sip] can SDP contain 2 'M'=20
lines<BR></FONT><BR></DIV>
<DIV></DIV>Hi All,<BR><BR>In SDP can we have two 'M' lines for re-INVITE =
i.e.=20
for call hold.<BR><BR>if yes what is RFC / draft<BR><BR>Any response is =
highly=20
appreciated<BR><BR>Regards<BR>Sudhir </BODY></HTML>

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_______________________________________________
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References: <5D1A7985295922448D5550C94DE2918002061E8B@DEEXC1U01.de.lucent.com><E6C2E8958BA59A4FB960963D475F7AC30BE3B850A2@mail.acmepacket.com>	<0D5F89FAC29E2C41B98A6A762007F5D0D6DEA6@GBNTHT12009MSX.gb002.siemens.net>	<C84E0A4ABA6DD74DA5221E0833A35DF30ABBDAB4@esebe101.NOE.Nokia.com><E6C2E8958BA59A4FB960963D475F7AC30EED5403CA@mail.acmepacket.com><4862A324.6020703@zonnet.nl><CA9998CD4A020D418654FCDEF4E707DF06DBE199@esealmw113.eemea.ericsson.se><4862BEE4.502@zonnet.nl><E6C2E8958BA59A4FB960963D475F7AC30EED54080B@mail.acmepacket.com><48633C02.2070703@zonnet.nl><CA9998CD4A020D418654FCDEF4E707DF06DBE53D@esealmw113.eemea.ericsson.se><4863441E.1090704@zonnet.nl><CA9998CD4A020D418654FCDEF4E707DF06DBE633@esealmw113.eemea.ericsson.se><48634AC8.30201@zonnet.nl>
	<CA9998CD4A020D418654FCDEF4E707DF06DE1641@esealmw113.eemea.ericsson.se>
From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Christer Holmberg" <christer.holmberg@ericsson.com>,
	"Jeroen van Bemmel" <jbemmel@zonnet.nl>
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Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use case
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Hi,

Another thing.

As currently defined in the draft, when the "keep" parameter is used for no=
n-registration calls (no matter whether it's UA-to-proxy or proxy-to-proxy)=
 the duration is only for the lifetime of the call.

When the "keep" parameter is used for registration calls, the duration is f=
or the lifetime of the registration (as for Outbound).

Regards,

Christer
 =


-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Chris=
ter Holmberg
Sent: 26. kes=E4kuuta 2008 11:06
To: Jeroen van Bemmel
Cc: sip@ietf.org; drage@alcatel-lucent.com; john.elwell@siemens.com; Markus=
.Isomaki@nokia.com
Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use case


Hi, =


>We're mixing different cases here. The 3GPP non-register emergency call =

>scenario is for an endpoint, I was talking about the proxy-to-proxy =

>case (I believe the 3GPP case can be addressed using standard mechanisms a=
vailable today, it's only about authorization policies and identification o=
f what calls are emergency calls).
>
>It doesn't make sense for a proxy to keep-alive connections with =

>arbitrary other proxies. Either it has a limited set of peers (such as =

>in case of an IP-PBX peering with a service provider), or it routes using =
DNS based on the domain in the request URI (or top Route header).
>For the latter case, the act of sending the INVITE itself will detect reac=
hability.
>
>So I suggest to limit the proxy-to-proxy use-case for keep-alive to =

>proxies that route towards a limited set of next hops (instead of DNS =

>on domain). According to RFC3261, such proxies would insert a Route =

>header (with loose routing)

Ok. Thanks for the clarification.

So, if I understand you correctly you either:

1. Propose that proxy-to-proxy should not be part of the scope of the draft=
 2. Proxy-to-proxy could be part of the scope of the draft, but we would no=
t necessarily use the same mechanism for UA-to-proxy and proxy-to-proxy.

Correct?

Regards,

Christer



Christer Holmberg wrote:
> Hi,
>
>   =

>> The only solution I can think of that would give less processing, =

>> would be to simply open a TCP connection (or UDP flow with STUN) =

>> without sending any request. Less "overkill", but it would open up a =

>> bunch of security issues ( e.g. TCP connection resources being =

>> consumed by unauthorized elements, leading to DoS; all the concerns with=
 STUNning before checking support, etc. ). So I don't see how sending at le=
ast one SIP request could be avoided. REGISTER is as good as any.
>>
>> Which proxies to register with: that would of course have to be =

>> provisioned somehow, this is no different from having to provision =

>> which outbound proxy to use
>>     =

>
> There is a big difference. A UA will always send the requests via the out=
bound proxy, but a proxy can communicate with an unlimited number of other =
proxies. Those proxies are normally found (unless a service route exists) u=
sing DNS when the proxy receives a request to forward.
>
>   =

>> UEs that don't register: the idea is to make all such UEs register, =

>> so there'd be no such case left
>>     =

>
> As said earlier, at least 3GPP has a requirement for non-register emergen=
cy calls.
>
> Regards,
>
> Christer
>
>
>
>
>
> Christer Holmberg wrote:
>   =

>> Jeroen,
>>
>> I don't understand what benefit it would be to establish outbound regist=
rations between all SIP entities that want to use the keep-alive/heartbeat.=
 Yes, we would "re-use an existing mechanism", but in my opinion it would r=
eally be an overkill,a hack - and anything but keep-it-simple...
>>
>> Also, how would the proxies know which proxies it should establish regis=
trations which?
>>
>> Also, it wouldn't work in the use-case with UEs that don't register.
>>
>> Regards,
>>
>> Christer
>>
>>
>> -----Original Message-----
>> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
>> Sent: 26. kes=E4kuuta 2008 9:50
>> To: Hadriel Kaplan
>> Cc: Christer Holmberg; Markus.Isomaki@nokia.com; =

>> john.elwell@siemens.com; drage@alcatel-lucent.com; sip@ietf.org
>> Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy =

>> use case
>>
>> Hadriel,
>>
>> I did not intend to suggest to use the registration interval timer for k=
eep-alive, of course all the regular flow keep-alive mechanisms of outbound=
 could be used.
>>
>> The benefit is that we would reuse the efforts put in outbound, instead =
of inventing yet another, only marginally different, mechanism for each spe=
cific case. From the sending element point of view, sending a REGISTER isn'=
t very different from sending an OPTIONS; subsequent keep-alives can then b=
ecome STUN or double CRLF, until the registration needs to be refreshed.
>>
>> Using REGISTER instead of OPTIONS also provides the means to use the =

>> standard authentication mechanisms, besides reusing outbound (and =

>> there's a few other reusable things, like Path headers, but let's =

>> keep it simple for the moment)
>>
>> Regards,
>> Jeroen
>>
>> Hadriel Kaplan wrote:
>>   =

>>     =

>>> Hey Jeroen,
>>> I'm not sure I follow you.  The SIP Forum SIP-Connect profile, and now =
TISPAN, make use of a REGISTER for an IP-PBX to register a whole set of AoR=
's to be routed to.  For those devices that need that ability, it makes sen=
se to use outbound of course, and its inherent keepalive, regardless of bei=
ng behind a NAT or not.  And I don't think Christer's suggesting otherwise.
>>>
>>> But for IP-PBX's or proxies which do not themselves represent an AoR ta=
rget of requests, or do not represent one in the domain of the device they =
want to perform keepalive with, why would we want to make them add such reg=
istration logic?  What would be the gain over just simply sending OPTIONS a=
t that point?
>>>
>>> The beauty of a STUN or double-CRLF keepalive is, in my mind: it's triv=
ial to construct, trivial to parse, very small and fixed size, can be separ=
ated or handled at a lower layer, explicit in its use, and does not get sto=
pped by a SIP-layer overload control.  In short: it's a transport connectio=
n-layer keepalive, no more no less.  And indicating it in the Via keeps it =
a hop-link thing, backwards-compatible, little or no provisioning, and with=
 no URI target addressing issues.
>>>
>>> -hadriel
>>>
>>>   =

>>>     =

>>>       =

>>>> -----Original Message-----
>>>> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
>>>> Sent: Wednesday, June 25, 2008 5:56 PM
>>>> To: Christer Holmberg
>>>> Cc: Hadriel Kaplan; Markus.Isomaki@nokia.com; =

>>>> john.elwell@siemens.com; drage@alcatel-lucent.com; sip@ietf.org
>>>> Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy =

>>>> use case
>>>>
>>>> No, not a B2BUA. Each element would have a virtual UA function "in =

>>>> parallel" to its regular function (e.g. being a proxy). And indeed, =

>>>> one way to implement this would be to keep either one (one way) or =

>>>> two (opposite ways) flows active between them.
>>>>
>>>> Like so:
>>>>
>>>> |-----------|                     |-----------|
>>>> |   Proxy     |                     |  Proxy     |
>>>> |-----------|                     |-----------|
>>>> |     UA      | <---------->|      UA     |
>>>> |-----------|                     |-----------|
>>>>
>>>> Each "UA" would implement both a simple registrar and a UAC =

>>>> performing registration. It could also be setup asymmetrically, =

>>>> with the (smaller) IP-PBX doing a single registration towards the
>>>> (bigger) Service provider network. Registration expiry would denote a =
loss of connectivity.
>>>>
>>>> Regards,
>>>> Jeroen
>>>>
>>>>
>>>> Christer Holmberg wrote:
>>>>     =

>>>>       =

>>>>         =

>>>>> Hi,
>>>>>
>>>>> So, you are proposing that each element should be a B2BUA, and =

>>>>> both
>>>>>       =

>>>>>         =

>>>>>           =

>>>> elements then register towards each other and use Outbound???
>>>>     =

>>>>       =

>>>>         =

>>>>> Regards,
>>>>>
>>>>> Christer
>>>>>
>>>>> -----Original Message-----
>>>>> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]
>>>>> Sent: 25. kes=E4kuuta 2008 22:57
>>>>> To: Hadriel Kaplan
>>>>> Cc: Markus.Isomaki@nokia.com; john.elwell@siemens.com;
>>>>> drage@alcatel-
>>>>>       =

>>>>>         =

>>>>>           =

>>>> lucent.com; sip@ietf.org; Christer Holmberg
>>>>     =

>>>>       =

>>>>         =

>>>>> Subject: Re: [Sip] Progress draft-holmberg-sip-keep: =

>>>>> proxy-to-proxy use
>>>>>       =

>>>>>         =

>>>>>           =

>>>> case
>>>>     =

>>>>       =

>>>>         =

>>>>> Hadriel, Markus,
>>>>>
>>>>> Instead of standardizing keep-alives between proxies, how about we
>>>>>       =

>>>>>         =

>>>>>           =

>>>> define a "virtual UA" on each element (similar to the one described =

>>>> in
>>>>     =

>>>>       =

>>>>         =

>>>>> RFC3261 section 16.7 point 6) to be used to provide this functionalit=
y?
>>>>> (using existing outbound functionality, perhaps both ways)
>>>>>
>>>>> Regards,
>>>>> Jeroen
>>>>>
>>>>> Hadriel Kaplan wrote:
>>>>>
>>>>>       =

>>>>>         =

>>>>>           =

>>>>>> Yes I am of that same opinion - that any real "IP-PBX" or =

>>>>>> whatever big enough NOT to be doing Registration, and to instead =

>>>>>> do static
>>>>>>         =

>>>>>>           =

>>>>>>             =

>>>> provisioning or DNS, would be given a static hole/DMZ address in =

>>>> their firewall/NAT.  But some of my customers have told me otherwise.
>>>> (interestingly mostly in APAC region) There's also some concern =

>>>> that while a static entry is there for inbound TCP connections, the =

>>>> PBX creates outbound ones to the service provider which are =

>>>> ephemeral port sources and need to live for very long durations =

>>>> (though why they can't just do TCP keepalive is beyond me, but I'm no =
expert).
>>>>     =

>>>>       =

>>>>         =

>>>>>> But anyway, the big issue we've seen is that we need both the PBX =

>>>>>> and
>>>>>>         =

>>>>>>           =

>>>>>>             =

>>>> the service provider box to detect failure before an active =

>>>> call/request attempt is made; to trigger alternate route selection =

>>>> without waiting for transport failure, and as a method to detect liven=
ess again and revert.
>>>> Today that's almost exclusively done with Options requests as far =

>>>> as I've seen, and lots of people don't seem to like that.
>>>>     =

>>>>       =

>>>>         =

>>>>>> -hadriel
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>         =

>>>>>>           =

>>>>>>             =

>>>>>>> -----Original Message-----
>>>>>>> From: Markus.Isomaki@nokia.com [mailto:Markus.Isomaki@nokia.com]
>>>>>>> Sent: Wednesday, June 25, 2008 3:11 PM
>>>>>>> To: john.elwell@siemens.com; Hadriel Kaplan; =

>>>>>>> drage@alcatel-lucent.com; sip@ietf.org
>>>>>>> Cc: christer.holmberg@ericsson.com
>>>>>>> Subject: RE: [Sip] Progress draft-holmberg-sip-keep: =

>>>>>>> proxy-to-proxy use case
>>>>>>>
>>>>>>> Hi,
>>>>>>>
>>>>>>> I'm a bit sceptical about the need for keep-alives between proxies.
>>>>>>> It is of course entirely possible that an enterprise PBX is =

>>>>>>> connected to (or peering with) a service provider proxy through =

>>>>>>> a NAT and/or a firewall. However, wouldn't such a NAT or =

>>>>>>> firewall be under the administration of either the enterprise =

>>>>>>> itself or its ISP (who quite often would be the SIP service =

>>>>>>> provider), and the required port forwardings or firewall rules coul=
d be set through administration.
>>>>>>> This means that there would not be need for keepalive traffic to =

>>>>>>> implicitely keep the mapping/pinhole open.
>>>>>>>
>>>>>>> Or are there really deployment cases where there are SIP PBXs =

>>>>>>> behind unadministrated NATs or firewalls?
>>>>>>>
>>>>>>> Wouldn't we then need keepalives for SMTP as well, or how has =

>>>>>>> the e-mail infrastructure managed to solve this problem?
>>>>>>>
>>>>>>> Markus
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>           =

>>>>>>>             =

>>>>>>>               =

>>>>>> _______________________________________________
>>>>>> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>>>>>> This list is for NEW development of the core SIP Protocol Use =

>>>>>> sip-implementors@cs.columbia.edu for questions on current sip Use =

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

>>>>>>           =

>>>>>>             =

>>>>>       =

>>>>>         =

>>>>>           =

>>>   =

>>>     =

>>>       =

>>   =

>>     =

>
>   =

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This list is for NEW development of the core SIP Protocol Use sip-implement=
ors@cs.columbia.edu for questions on current sip Use sipping@ietf.org for n=
ew developments on the application of sip
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Cc: sip@ietf.org, Paul Kyzivat <pkyzivat@cisco.com>, Dan Wing <dwing@cisco.com>
Subject: Re: [Sip] Toward the Evolution of SIP and Related Working Groups
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Comments inline:

Elwell, John wrote:
> Hannes, 
>
>   
>> -----Original Message-----
>> From: Tschofenig, Hannes (NSN - FI/Espoo) 
>> [mailto:hannes.tschofenig@nsn.com] 
>> Sent: 25 June 2008 14:00
>> To: Elwell, John; Paul Kyzivat
>> Cc: sip@ietf.org; Dan Wing
>> Subject: RE: [Sip] Toward the Evolution of SIP and Related 
>> Working Groups
>>
>>
>> Consider the following scenario:
>>
>>               +-----------+            +-----------+
>>               |SIP        |            |SIP        |
>>       +------>|Proxy      |<---------->|Proxy      |<------+
>>       |       |Server X   |   TLS      |Server Y   |       |
>>       |       +-----------+            +-----------+       |
>>       |                                                    |
>>       | TLS or                                      TLS or |
>>       | SIP Digest                              SIP Digest |
>>       |                                                    |
>>       |                                                    |
>>       v                                                    v
>>   +-----------+                                     +-----------+
>>   |SIP        |                                     |SIP        |
>>   |User Agent |          RTP                        |User Agent |
>>   |Alice      | <=================================> |Bob        |
>>   +-----------+                                     +-----------+
>>  
>>
>> When there are no further proxies between X and Y then Y has the
>> information that Alice was authenticated by X. Proxy Y would pass that
>> info on to Bob. Bob trusts Y.
>>
>> Obviously, the two VoIP providers may have a far more complicated
>> infrastructure with multiple proxies but they all belong to the same
>> domain and could be seen from external as just one box. 
>>
>> The same would not work when there are more proxies between X and Y.
>> However, these guys that prefer such a deployment belong more to the
>> chain of trust camp rather than the end-to-end / email alike peering
>> camp. It is rather unlikely that you get their support for getting SIP
>> Identity to work in their networks.
>>
>> So, do we have an indication that some folks plan to use SIP Identity
>> for their deployment? 
>>     
> [JRE] Well, what about enterprise via service provider to enterprise? As
> an enterprise provider, I would like to be able to use SIP identity, but
> a precondition is that it gets through that intermediate service
> provider(s).
>
>   
Why would like to pick a service provider in the first place? Why are 
they deploying B2BUAs?

If you really want todo that then pick one that does not damage your 
security. You are the customer and you decide.

If you want to pick such as service provider then why does the chain of 
trust not work for you?  Hopefully you trust the provider you have chosen.

.... just trying to figure out what the typical deployment cases are and 
why we don't see any of this in XMPP.

>   
>> Wouldn't it be better to rely on something like SIP CERT for a better
>> end-to-end security mechanism (ignoring for a moment that SIP 
>> CERT also
>> uses SIP Identity for "simplified deployment" reasons whereby 
>> one has to
>> state that the usage of SIP Identity for SUBSCRIBE/NOTIFY might have
>> different B2BUA aspects). 
>>     
> [JRE] You have lost me there. What is SIP CERT? Can you point me at a
> draft?
>   
http://tools.ietf.org/html/draft-ietf-sip-certs-06

Ciao
Hannes

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

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Date: Thu, 26 Jun 2008 10:10:34 +0100
From: "Elwell, John" <john.elwell@siemens.com>
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Subject: Re: [Sip] Toward the Evolution of SIP and Related Working Groups
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Hannes,
 

> -----Original Message-----
> From: Hannes Tschofenig [mailto:Hannes.Tschofenig@gmx.net] 
> Sent: 26 June 2008 09:26
> To: Elwell, John
> Cc: Tschofenig, Hannes (NSN - FI/Espoo); Paul Kyzivat; 
> sip@ietf.org; Dan Wing
> Subject: Re: [Sip] Toward the Evolution of SIP and Related 
> Working Groups
> 
> Comments inline:
> 
> Elwell, John wrote:
> > Hannes, 
> >
> >   
> >> -----Original Message-----
> >> From: Tschofenig, Hannes (NSN - FI/Espoo) 
> >> [mailto:hannes.tschofenig@nsn.com] 
> >> Sent: 25 June 2008 14:00
> >> To: Elwell, John; Paul Kyzivat
> >> Cc: sip@ietf.org; Dan Wing
> >> Subject: RE: [Sip] Toward the Evolution of SIP and Related 
> >> Working Groups
> >>
> >>
> >> Consider the following scenario:
> >>
> >>               +-----------+            +-----------+
> >>               |SIP        |            |SIP        |
> >>       +------>|Proxy      |<---------->|Proxy      |<------+
> >>       |       |Server X   |   TLS      |Server Y   |       |
> >>       |       +-----------+            +-----------+       |
> >>       |                                                    |
> >>       | TLS or                                      TLS or |
> >>       | SIP Digest                              SIP Digest |
> >>       |                                                    |
> >>       |                                                    |
> >>       v                                                    v
> >>   +-----------+                                     +-----------+
> >>   |SIP        |                                     |SIP        |
> >>   |User Agent |          RTP                        |User Agent |
> >>   |Alice      | <=================================> |Bob        |
> >>   +-----------+                                     +-----------+
> >>  
> >>
> >> When there are no further proxies between X and Y then Y has the
> >> information that Alice was authenticated by X. Proxy Y 
> would pass that
> >> info on to Bob. Bob trusts Y.
> >>
> >> Obviously, the two VoIP providers may have a far more complicated
> >> infrastructure with multiple proxies but they all belong 
> to the same
> >> domain and could be seen from external as just one box. 
> >>
> >> The same would not work when there are more proxies 
> between X and Y.
> >> However, these guys that prefer such a deployment belong 
> more to the
> >> chain of trust camp rather than the end-to-end / email 
> alike peering
> >> camp. It is rather unlikely that you get their support for 
> getting SIP
> >> Identity to work in their networks.
> >>
> >> So, do we have an indication that some folks plan to use 
> SIP Identity
> >> for their deployment? 
> >>     
> > [JRE] Well, what about enterprise via service provider to 
> enterprise? As
> > an enterprise provider, I would like to be able to use SIP 
> identity, but
> > a precondition is that it gets through that intermediate service
> > provider(s).
> >
> >   
> Why would like to pick a service provider in the first place? Why are 
> they deploying B2BUAs?
> 
> If you really want todo that then pick one that does not damage your 
> security. You are the customer and you decide.
> 
> If you want to pick such as service provider then why does 
> the chain of 
> trust not work for you?  Hopefully you trust the provider you 
> have chosen.
[JRE] I might trust my service provider, but I might not trust (or even
know of) the peer enterprise's service provider. And what if the two
service providers use indirect peering rather than direct peering? That
is the problem, there can be just too many domains along the path of a
call, particularly a call that gets forwarded. It seems too hard to
figure out for every possible scenario how to get hop-by-hop to work,
when an end-domain-to-end-domain solution would always work if there is
no interference from intermediate domains.

> 
> .... just trying to figure out what the typical deployment 
> cases are and 
> why we don't see any of this in XMPP.
> 
> >   
> >> Wouldn't it be better to rely on something like SIP CERT 
> for a better
> >> end-to-end security mechanism (ignoring for a moment that SIP 
> >> CERT also
> >> uses SIP Identity for "simplified deployment" reasons whereby 
> >> one has to
> >> state that the usage of SIP Identity for SUBSCRIBE/NOTIFY 
> might have
> >> different B2BUA aspects). 
> >>     
> > [JRE] You have lost me there. What is SIP CERT? Can you 
> point me at a
> > draft?
> >   
> http://tools.ietf.org/html/draft-ietf-sip-certs-06
[JRE] I don't understand - how does sip-certs provide end-to-end
authentication? It might help me to locate a certificate that is used in
some authentication mechanism such as Identity, but it doesn't itself
provide an authentication mechanism.

John

> 
> Ciao
> Hannes
> 
> > John
> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >   
> 
> 
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From: "Alfred E. Heggestad" <aeh@db.org>
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Subject: Re: [Sip] can SDP contain 2 'M' lines
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Sudhir Kumar Reddy wrote:
> Hi All,
> 
> In SDP can we have two 'M' lines for re-INVITE i.e. for call hold.
> 
> if yes what is RFC / draft
> 
> Any response is highly appreciated
> 

yes, please see details here:

   http://www.ietf.org/rfc/rfc4566.txt

   http://www.ietf.org/rfc/rfc3264.txt


/alfred
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To: "Elwell, John" <john.elwell@siemens.com>
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>> http://tools.ietf.org/html/draft-ietf-sip-certs-06
>>     
> [JRE] I don't understand - how does sip-certs provide end-to-end
> authentication? It might help me to locate a certificate that is used in
> some authentication mechanism such as Identity, but it doesn't itself
> provide an authentication mechanism.
>
> John

Two of the current proposals to fix SIP Identity combine it with 
DTLS-SRTP. An alternative is to use SIP CERT with DTLS-SRTP.

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Hannes Tschofenig wrote:

> If you want to pick such as service provider then why does the chain of 
> trust not work for you?  Hopefully you trust the provider you have chosen.

Well, that seems obvious. If the SP is willing to trust *me*, then it is 
probably willing to trust its other customers. If I am not willing to 
trust its other customers, then I can't trust it.

	Paul
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Hannes,


> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of Hannes Tschofenig
> Sent: 26 June 2008 11:26
> To: Elwell, John
> Cc: sip@ietf.org; Paul Kyzivat; Dan Wing
> Subject: Re: [Sip] Toward the Evolution of SIP and Related 
> Working Groups
> 
> 
> >> http://tools.ietf.org/html/draft-ietf-sip-certs-06
> >>     
> > [JRE] I don't understand - how does sip-certs provide end-to-end
> > authentication? It might help me to locate a certificate 
> that is used in
> > some authentication mechanism such as Identity, but it 
> doesn't itself
> > provide an authentication mechanism.
> >
> > John
> 
> Two of the current proposals to fix SIP Identity combine it with 
> DTLS-SRTP. An alternative is to use SIP CERT with DTLS-SRTP.
[JRE] So you propose having certificates in the UAs, and have these
deployed using sip-certs?
I wonder whether this meets the requirement:
"R-CERTS:
         The key management protocol MUST NOT require that end-users
         obtain credentials (certificates or private keys) from a third-
         party trust anchor."
Does the credential server count as a third party trust anchor? If so,
we would not meet this requirement. But assuming the server is in the
same domain as the UAs it supplies credentials to, it is probably not
considered third party and probably meets the spirit of the requirement.

Also, I think there would need to be some adaptation of sip-certs. For
example, sip-certs explicitly says it is for use with S/MIME. Also, as
you pointed out, this too relies on RFC 4474 (but in a NOTIFY). Maybe
HTTPS would be more appropriate for this part.

If Bob has several UAs, each would receive the same private key and cert
from the credentials server. I don't know whether this has any
implications for DTLS-SRTP (c.f., self-signed certs, which would
doubtless be different in each UA).

Similarly to some of the other alternatives, this would provide an
authenticated identity for DTLS-SRTP, but not for calls that don't use
DTLS-SRTP, and also not for SIP requests not related to calls such as
MESSAGE, SUBSCRIBE, NOTIFY, and also not for mid-dialog requests that
don't include SDP (where the certificate fingerprint resides), such as
BYE, offerless UPDATE, the dreaded INFO.

Intermediate domains would need to pass SUBSCRIBE/NOTIFY requests - this
might be almost as big an ask as requiring them not to interfere with an
RFC 4474 certificate. Alternatively, perhaps enterprises etc. that
insist on intermediaries for calls might open their doors to direct
peering for SUBSCRIBE/NOTIY.

I can still see issues with the actual identity. For example, with an
E.164 number, I send +1234@mydomain.com in From or PAI, but by the time
it reaches you it has been changed to +1234@serviceprovider.net, so your
UA would send the SUBSCRIBE request to +1234@mydomain.com, and that
would somehow need to reach +1234@mydomain.com.

But yes, this is an alternative approach worthy of further exploration.

John
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From: "Tschofenig, Hannes (NSN - FI/Espoo)" <hannes.tschofenig@nsn.com>
To: "ext Paul Kyzivat" <pkyzivat@cisco.com>,
	"Hannes Tschofenig" <Hannes.Tschofenig@gmx.net>
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Cc: sip@ietf.org, "Elwell, John" <john.elwell@siemens.com>,
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Subject: [Sip] SIP Identity
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[changed the title since we drifted a bit away from the original
question] 

I am trying to figure out where the problems are: 

The SIPit 21 report 
http://www.ietf.org/mail-archive/web/sip/current/msg21291.html
indicated that SIP Identity was available with 12% of the
implementations. (Not sure what that actually means since one would have
to look at the details.)

The SIPit 22 report 
http://www.ietf.org/mail-archive/web/sip/current/msg23285.html
says 16% identity (But there were no Identity servers at the event)

Maybe this means that nobody showed up that has implemented the
Authentication Service but I hope that someone has implemented it
somewhere. 

~15 % is not a large number; but it's a start. 

Now, it would be interesting to hear whether these guys have actually
ever turned it on and used it. What was their experience? Did they
indeed encounter problems with certain SBC configurations? 

Would more folks use it when some of the weaknesses on the E.164 number
usage are documented?

Would more folks use it when combined with DTLS-SRTP (as proposed in two
documents)?

How would a change to the signature algorithm impact the situation?

Ciao
Hannes

>-----Original Message-----
>From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
>Behalf Of ext Paul Kyzivat
>Sent: 26 June, 2008 15:36
>To: Hannes Tschofenig
>Cc: sip@ietf.org; Elwell, John; Dan Wing
>Subject: Re: [Sip] Toward the Evolution of SIP and Related 
>Working Groups
>
>
>
>Hannes Tschofenig wrote:
>
>> If you want to pick such as service provider then why does the chain 
>> of trust not work for you?  Hopefully you trust the provider 
>you have chosen.
>
>Well, that seems obvious. If the SP is willing to trust *me*, 
>then it is probably willing to trust its other customers. If I 
>am not willing to trust its other customers, then I can't trust it.
>
>	Paul
>_______________________________________________
>Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol Use 
>sip-implementors@cs.columbia.edu for questions on current sip 
>Use sipping@ietf.org for new developments on the application of sip
>
_______________________________________________
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John Elwell wrote: 

> I can still see issues with the actual identity. For example, with an
> E.164 number, I send +1234@mydomain.com in From or PAI, but 
> by the time it reaches you it has been changed to 
> +1234@serviceprovider.net, so your
> UA would send the SUBSCRIBE request to +1234@mydomain.com, and that
> would somehow need to reach +1234@mydomain.com.

Which brings us back into draft-wing-sip-e164-rrc-01.txt territory.
(Not a bad thing, in my mind; I just wanted to point it out).

-d

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Hi,

   Is there any Softphone client which sends media attributes: 'a= inactive'
   when the call is put on hold?
   Tried SJ phone, Xlite, BT softphone and all of these send 'a=sendonly'.
   The client should be capable of reg with SER. This is to test the msg
   handling by a SBC, so SIPp can't be used as i need to check the RTP flow
as
   well.

   Cheers,
   Vishnu

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<div>Hi,</div>
<div>&nbsp;</div>
<div>&nbsp;&nbsp; Is there any Softphone client which sends media attributes: &#39;a= inactive&#39;<br>&nbsp; &nbsp;when the call is put on hold?<br>&nbsp; &nbsp;Tried SJ phone, Xlite, BT softphone and all of these send &#39;a=sendonly&#39;.<br>&nbsp; &nbsp;The client should be capable of reg with SER. This is to test the msg<br>
&nbsp; &nbsp;handling by a SBC, so SIPp can&#39;t be used as i need to check the RTP flow as<br>&nbsp; &nbsp;well.</div>
<div><br>&nbsp;&nbsp;&nbsp;Cheers,<br>&nbsp; &nbsp;Vishnu<br>&nbsp;</div>

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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
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Dear all,
at the moment I'm looking for comments on the =
http://tools.ietf.org/id/draft-jesske-sipping-etsi-ngn-reason-03.txt, to =
progress the work on that.=20
The reason in responses will be used within the IMS so progress of this =
draft is needed.
=20
Also I've got the comment that the work should be shifted to SIP WG.
=20
Thoughts? Comments?
=20
Best Regards
=20
Roland
=20
 <http://tools.ietf.org/id/draft-jesske-sipping-etsi-ngn-reason-03.txt>  =

=20

Deutsche Telekom AG=20
Zentrum Technik Einf=FChrung=20
Roland Jesske=20
Gateways, Protokolle, Konvergenz, TE32-1
Heinrich-Hertz-Str. 3-7,  64295 Darmstadt, =20
Postfach, 64307 Darmstadt (Postanschrift)=20
+496151 628 2766 (Tel)
+491718618445  (Mobile)=20
E-Mail: r.jesske@telekom.de=20

 <http://www.telekom.de/> http://www.telekom.de=20

Registerrechtliche Unternehmensangaben:=20
Deutsche Telekom AG
Aufsichtsrat: Prof. Dr Ulrich Lehner (Vorsitzender)
Vorstand: Ren=E9 Obermann (Vorsitzender),
Dr. Karl-Gerhard Eick (stellvertretender Vorsit-zender),
Hamid Akhavan, Reinhard Clemens, Timotheus H=F6ttges, Thomas =
Sattelberger=20
Handelsregister: Amtsgericht Bonn HRB 6794,=20
Sitz der Gesellschaft: Bonn=20
WEEE-Reg.-Nr.: DE50478376=20
***************************************************************=20
Deutsche Telekom AG=20
Supervisory Board: Prof. Dr Ulrich Lehner (Chairman)=20
Board of Management: Ren=E9 Obermann (Chairman),=20
Dr. Karl-Gerhard Eick (Deputy Chairman),=20
Hamid Akhavan, Reinhard Clemens, Timotheus H=F6ttges,  Thomas =
Sattelberger=20
Commercial register: Amtsgericht Bonn HRB 6794,=20
corporate seat: Bonn=20
WEEEreg.no: DE50478376=20

=20

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charset=3Diso-8859-1">


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<BODY>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D043404315-26062008>Dear=20
all,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D043404315-26062008>at the =
moment I'm=20
looking for comments on the <A=20
href=3D"http://tools.ietf.org/id/draft-jesske-sipping-etsi-ngn-reason-03.=
txt">http://tools.ietf.org/id/draft-jesske-sipping-etsi-ngn-reason-03.txt=
</A>,=20
to progress the work on that.&nbsp;</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D043404315-26062008>The =
reason in=20
responses will be used within the IMS so&nbsp;progress of this draft is=20
needed.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D043404315-26062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D043404315-26062008>Also =
I've got the=20
comment that the work should be shifted to SIP WG.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D043404315-26062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN =
class=3D043404315-26062008>Thoughts?=20
Comments?</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D043404315-26062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D043404315-26062008>Best=20
Regards</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D043404315-26062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D043404315-26062008>Roland</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D043404315-26062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><A=20
href=3D"http://tools.ietf.org/id/draft-jesske-sipping-etsi-ngn-reason-03.=
txt"></A></FONT>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<P align=3Dleft><SPAN lang=3Dde><FONT face=3DArial size=3D1>Deutsche =
Telekom=20
AG</FONT></SPAN>&nbsp;<BR><SPAN lang=3Dde><FONT face=3DArial =
size=3D1>Zentrum Technik=20
Einf=FChrung</FONT></SPAN> <BR><SPAN lang=3Dde><FONT face=3DArial =
size=3D1>Roland=20
Jesske</FONT></SPAN> <BR><SPAN lang=3Dde><FONT face=3DArial =
size=3D1>Gateways,=20
Protokolle, Konvergenz, TE32-1</FONT></SPAN><BR><SPAN lang=3Dde><FONT =
size=3D1><FONT=20
face=3DArial><SPAN lang=3Dde><B>Heinrich-Hertz-Str. =
3-7,</B>&nbsp;</SPAN> 64295=20
Darmstadt,&nbsp;</FONT></FONT></SPAN> <BR><SPAN lang=3Dde><FONT =
face=3DArial=20
size=3D1>Postfach, 64307 Darmstadt =
(Postanschrift)</FONT></SPAN>&nbsp;<BR><FONT=20
face=3DArial size=3D1>+496151 628 2766 (Tel)</FONT><BR><SPAN =
lang=3Dfr><FONT=20
face=3DArial size=3D1>+491718618445 &nbsp;(Mobile)</FONT></SPAN><SPAN=20
lang=3Dde></SPAN> <BR><SPAN lang=3Dfr><FONT face=3DArial =
size=3D1>E-Mail: <A=20
href=3D"mailto:r.jesske@telekom.de">r.jesske@telekom.de</A> =
</FONT></SPAN></P>
<P align=3Dleft><SPAN lang=3Dfr></SPAN><SPAN lang=3Dfr><A=20
href=3D"http://www.telekom.de/"><FONT face=3DArial=20
size=3D2>http://www.telekom.de</FONT></A><FONT face=3DArial =
color=3D#0000ff size=3D2>=20
</FONT></SPAN><SPAN lang=3Dde></P></SPAN><SPAN lang=3Dde></SPAN><SPAN =
lang=3Dde>
<P class=3Deinzug align=3Dleft></SPAN><FONT size=3D1><FONT =
face=3DArial><SPAN=20
lang=3Dde><U>Registerrechtliche Unternehmensangaben:</U></SPAN>=20
<BR></FONT></FONT><SPAN lang=3Dde><FONT size=3D1><FONT =
face=3DArial>Deutsche Telekom=20
AG<BR>Aufsichtsrat:&nbsp;Prof. Dr Ulrich Lehner<SPAN=20
class=3D012202609-06032008><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: black; FONT-FAMILY: Arial; =
mso-fareast-font-family: 'Times New Roman'; mso-ansi-language: DE; =
mso-fareast-language: EN-US; mso-bidi-language: AR-SA">=20
</SPAN></SPAN></FONT></FONT><SPAN class=3D012202609-06032008><FONT =
face=3DArial=20
size=3D1>(Vorsitzender)<BR></FONT></SPAN></SPAN><SPAN lang=3Dde><FONT =
face=3DArial=20
size=3D1>Vorstand: </FONT><FONT face=3DArial size=3D1>Ren=E9 Obermann=20
(Vorsitzender),<BR></FONT></SPAN><FONT size=3D1><FONT face=3DArial><SPAN =
lang=3Dde>Dr.=20
Karl-Gerhard Eick (stellvertretender Vorsit-zender),<BR>Hamid Akhavan, =
Reinhard=20
Clemens, Timotheus H=F6ttges, Thomas Sattelberger <BR></SPAN><SPAN=20
lang=3Dde>Handelsregister: Amtsgericht Bonn HRB 6794,</SPAN> <BR><SPAN=20
lang=3Dde>Sitz der Gesellschaft: Bonn</SPAN> <BR><SPAN =
lang=3Dde>WEEE-Reg.-Nr.:=20
DE50478376</SPAN> <BR><SPAN=20
lang=3Dde>***************************************************************=
</SPAN>=20
<BR><SPAN lang=3Dde>Deutsche Telekom AG</SPAN> <BR><SPAN =
lang=3Den-gb>Supervisory=20
Board: <SPAN class=3D012202609-06032008>Prof. Dr Ulrich Lehner=20
</SPAN>(Chairman)</SPAN> <BR><SPAN lang=3Den-gb>Board of Management: =
Ren=E9 Obermann=20
(Chairman), </SPAN><BR><SPAN lang=3Den-gb>Dr. Karl-Gerhard Eick (Deputy=20
Chairman),</SPAN> <BR><SPAN lang=3Den-gb>Hamid Akhavan, Reinhard =
Clemens,=20
</SPAN><SPAN lang=3Dde>Timotheus H=F6ttges,&nbsp; Thomas =
Sattelberger</SPAN><SPAN=20
lang=3Den-gb></SPAN><SPAN lang=3Den-gb></SPAN> <BR><SPAN =
lang=3Dde>Commercial=20
register: Amtsgericht Bonn HRB 6794,</SPAN> <BR><SPAN =
lang=3Dde>corporate seat:=20
Bonn</SPAN> <BR><SPAN lang=3Dde>WEEEreg.no: =
DE50478376</SPAN></FONT></FONT> </P>
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_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============0153206829==--


From sip-bounces@ietf.org  Thu Jun 26 09:31:26 2008
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From: "Mary Barnes" <mary.barnes@nortel.com>
To: "Jesske, R" <R.Jesske@telekom.de>, <sipping@ietf.org>, <sip@ietf.org>
Subject: Re: [Sip] [Sipping] Progress on
	draft-jesske-sipping-etsi-ngn-reason-03
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If the requirements in this document are agreed in SIPPING WG (the doc =
seems really sparse in terms of requirements at this point), THEN the =
work could progress to the SIP WG. But, we first need to agree =
requirements.=20
=20
I see your applicability section and earlier there's reference to the =
TISPAN requirements document (which is expired). Can you pull the =
specific requirements from that document into this one or at least let's =
discuss on the list?  In scanning that requirements document, it's not =
clear to me which requirement this header is intended to meet. If it is =
intended to support many of those simulation services, then that should =
be stated. And, so I believe this brings us to the question as to =
whether we are going to support those simulation services?=20
=20
Given that many of those services are being developed in BLISS, would it =
not be prudent to wait until we see the details of those services to =
suggest that we need this header?  Or has that already been discussed =
and I've missed it? If so, please provide us a pointer to that =
discussion.  In that case, we could agree that SIPPING WG does not need =
to spend much time on the requirements and we could more easily agree =
that this could move to SIP WG to be completed.  Or, are there other =
requirements for this header? =20
=20
And, just for clarification, are their plans to progress that other =
general requirements document or should the WG treat that document as we =
did in the past the 3GPP requirements document as a working document, =
from which to pull specific requirements and solution proposals?  My =
inclination would be the latter, but we need feedback from the WG =
(including ETSI/TISPAN/IMS folks, of course).=20
=20
And, just a note, I bcc'ed SIP WG on this thread, since I would prefer =
this thread stay only in SIPPING WG for now, in particular just to keep =
down the volume on the SIP list, which is already really busy.
=20
Thanks,
Mary.=20

________________________________

From: sipping-bounces@ietf.org [mailto:sipping-bounces@ietf.org] On =
Behalf Of Jesske, R
Sent: Thursday, June 26, 2008 10:57 AM
To: sipping@ietf.org; sip@ietf.org
Subject: [Sipping] Progress on draft-jesske-sipping-etsi-ngn-reason-03


Dear all,
at the moment I'm looking for comments on the =
http://tools.ietf.org/id/draft-jesske-sipping-etsi-ngn-reason-03.txt, to =
progress the work on that.=20
The reason in responses will be used within the IMS so progress of this =
draft is needed.
=20
Also I've got the comment that the work should be shifted to SIP WG.
=20
Thoughts? Comments?
=20
Best Regards
=20
Roland
=20
<http://tools.ietf.org/id/draft-jesske-sipping-etsi-ngn-reason-03.txt> =20
=20

Deutsche Telekom AG=20
Zentrum Technik Einf=FChrung=20
Roland Jesske=20
Gateways, Protokolle, Konvergenz, TE32-1
Heinrich-Hertz-Str. 3-7,  64295 Darmstadt, =20
Postfach, 64307 Darmstadt (Postanschrift)=20
+496151 628 2766 (Tel)
+491718618445  (Mobile)=20
E-Mail: r.jesske@telekom.de=20

http://www.telekom.de <http://www.telekom.de/> =20

Registerrechtliche Unternehmensangaben:=20
Deutsche Telekom AG
Aufsichtsrat: Prof. Dr Ulrich Lehner (Vorsitzender)
Vorstand: Ren=E9 Obermann (Vorsitzender),
Dr. Karl-Gerhard Eick (stellvertretender Vorsit-zender),
Hamid Akhavan, Reinhard Clemens, Timotheus H=F6ttges, Thomas =
Sattelberger=20
Handelsregister: Amtsgericht Bonn HRB 6794,=20
Sitz der Gesellschaft: Bonn=20
WEEE-Reg.-Nr.: DE50478376=20
***************************************************************=20
Deutsche Telekom AG=20
Supervisory Board: Prof. Dr Ulrich Lehner (Chairman)=20
Board of Management: Ren=E9 Obermann (Chairman),=20
Dr. Karl-Gerhard Eick (Deputy Chairman),=20
Hamid Akhavan, Reinhard Clemens, Timotheus H=F6ttges,  Thomas =
Sattelberger=20
Commercial register: Amtsgericht Bonn HRB 6794,=20
corporate seat: Bonn=20
WEEEreg.no: DE50478376=20

=20

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<BODY>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D625541316-26062008>If the requirements&nbsp;in this document are =
agreed in=20
SIPPING WG (the doc seems really sparse in terms of requirements at this =
point),=20
THEN the work could progress to the SIP WG. But, we first need to agree=20
requirements. </SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D625541316-26062008></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D625541316-26062008>I see your applicability section and earlier =
there's=20
reference to the TISPAN requirements document (which is expired). Can =
you pull=20
the specific requirements from that document into this one or at least =
let's=20
discuss on the list?&nbsp; In scanning that requirements document, it's =
not=20
clear to me which requirement this header is intended to meet. If it is =
intended=20
to support many of those simulation services, then that should be =
stated. And,=20
so I believe this brings us to the question as to whether we are going =
to=20
support those simulation services? </SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D625541316-26062008></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D625541316-26062008>Given that many of those services are being =
developed=20
in BLISS, would it not be prudent to wait until we see the details of =
those=20
services to suggest that we need this header?&nbsp; Or has that already =
been=20
discussed and I've missed it? If so, please provide us a pointer to that =

discussion.&nbsp; In that case, we could agree that SIPPING WG does not =
need to=20
spend much time on the requirements and we could more easily agree that =
this=20
could move to SIP WG to be completed.&nbsp; Or, are there other =
requirements for=20
this header?&nbsp; </SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D625541316-26062008></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D625541316-26062008>And, just for clarification, are their plans =
to=20
progress that other general requirements document or should the WG treat =
that=20
document as we did in the past the 3GPP requirements document as a =
working=20
document, from which to pull specific requirements and solution =
proposals?&nbsp;=20
My inclination would be the latter, but we need feedback from the WG =
(including=20
ETSI/TISPAN/IMS folks, of course). </SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D625541316-26062008></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D625541316-26062008>And, just a note, I bcc'ed SIP WG on this =
thread, since=20
I would prefer this thread stay only in SIPPING WG for now, in =
particular just=20
to keep down the volume on the SIP list, which is already really=20
busy.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D625541316-26062008></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D625541316-26062008>Thanks,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D625541316-26062008>Mary. </SPAN></FONT></DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> sipping-bounces@ietf.org=20
[mailto:sipping-bounces@ietf.org] <B>On Behalf Of </B>Jesske, =
R<BR><B>Sent:</B>=20
Thursday, June 26, 2008 10:57 AM<BR><B>To:</B> sipping@ietf.org;=20
sip@ietf.org<BR><B>Subject:</B> [Sipping] Progress on=20
draft-jesske-sipping-etsi-ngn-reason-03<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D043404315-26062008>Dear=20
all,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D043404315-26062008>at the =
moment I'm=20
looking for comments on the <A=20
href=3D"http://tools.ietf.org/id/draft-jesske-sipping-etsi-ngn-reason-03.=
txt">http://tools.ietf.org/id/draft-jesske-sipping-etsi-ngn-reason-03.txt=
</A>,=20
to progress the work on that.&nbsp;</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D043404315-26062008>The =
reason in=20
responses will be used within the IMS so&nbsp;progress of this draft is=20
needed.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D043404315-26062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D043404315-26062008>Also =
I've got the=20
comment that the work should be shifted to SIP WG.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D043404315-26062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN =
class=3D043404315-26062008>Thoughts?=20
Comments?</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D043404315-26062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D043404315-26062008>Best=20
Regards</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D043404315-26062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D043404315-26062008>Roland</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D043404315-26062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><A=20
href=3D"http://tools.ietf.org/id/draft-jesske-sipping-etsi-ngn-reason-03.=
txt"></A></FONT>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<P align=3Dleft><SPAN lang=3Dde><FONT face=3DArial size=3D1>Deutsche =
Telekom=20
AG</FONT></SPAN>&nbsp;<BR><SPAN lang=3Dde><FONT face=3DArial =
size=3D1>Zentrum Technik=20
Einf=FChrung</FONT></SPAN> <BR><SPAN lang=3Dde><FONT face=3DArial =
size=3D1>Roland=20
Jesske</FONT></SPAN> <BR><SPAN lang=3Dde><FONT face=3DArial =
size=3D1>Gateways,=20
Protokolle, Konvergenz, TE32-1</FONT></SPAN><BR><SPAN lang=3Dde><FONT =
size=3D1><FONT=20
face=3DArial><SPAN lang=3Dde><B>Heinrich-Hertz-Str. =
3-7,</B>&nbsp;</SPAN> 64295=20
Darmstadt,&nbsp;</FONT></FONT></SPAN> <BR><SPAN lang=3Dde><FONT =
face=3DArial=20
size=3D1>Postfach, 64307 Darmstadt =
(Postanschrift)</FONT></SPAN>&nbsp;<BR><FONT=20
face=3DArial size=3D1>+496151 628 2766 (Tel)</FONT><BR><SPAN =
lang=3Dfr><FONT=20
face=3DArial size=3D1>+491718618445 &nbsp;(Mobile)</FONT></SPAN><SPAN=20
lang=3Dde></SPAN> <BR><SPAN lang=3Dfr><FONT face=3DArial =
size=3D1>E-Mail: <A=20
href=3D"mailto:r.jesske@telekom.de">r.jesske@telekom.de</A> =
</FONT></SPAN></P>
<P align=3Dleft><SPAN lang=3Dfr></SPAN><SPAN lang=3Dfr><A=20
href=3D"http://www.telekom.de/"><FONT face=3DArial=20
size=3D2>http://www.telekom.de</FONT></A><FONT face=3DArial =
color=3D#0000ff size=3D2>=20
</FONT></SPAN><SPAN lang=3Dde></P></SPAN><SPAN lang=3Dde></SPAN><SPAN =
lang=3Dde>
<P class=3Deinzug align=3Dleft></SPAN><FONT size=3D1><FONT =
face=3DArial><SPAN=20
lang=3Dde><U>Registerrechtliche Unternehmensangaben:</U></SPAN>=20
<BR></FONT></FONT><SPAN lang=3Dde><FONT size=3D1><FONT =
face=3DArial>Deutsche Telekom=20
AG<BR>Aufsichtsrat:&nbsp;Prof. Dr Ulrich Lehner<SPAN=20
class=3D012202609-06032008><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: black; FONT-FAMILY: Arial; =
mso-fareast-font-family: 'Times New Roman'; mso-ansi-language: DE; =
mso-fareast-language: EN-US; mso-bidi-language: AR-SA">=20
</SPAN></SPAN></FONT></FONT><SPAN class=3D012202609-06032008><FONT =
face=3DArial=20
size=3D1>(Vorsitzender)<BR></FONT></SPAN></SPAN><SPAN lang=3Dde><FONT =
face=3DArial=20
size=3D1>Vorstand: </FONT><FONT face=3DArial size=3D1>Ren=E9 Obermann=20
(Vorsitzender),<BR></FONT></SPAN><FONT size=3D1><FONT face=3DArial><SPAN =
lang=3Dde>Dr.=20
Karl-Gerhard Eick (stellvertretender Vorsit-zender),<BR>Hamid Akhavan, =
Reinhard=20
Clemens, Timotheus H=F6ttges, Thomas Sattelberger <BR></SPAN><SPAN=20
lang=3Dde>Handelsregister: Amtsgericht Bonn HRB 6794,</SPAN> <BR><SPAN=20
lang=3Dde>Sitz der Gesellschaft: Bonn</SPAN> <BR><SPAN =
lang=3Dde>WEEE-Reg.-Nr.:=20
DE50478376</SPAN> <BR><SPAN=20
lang=3Dde>***************************************************************=
</SPAN>=20
<BR><SPAN lang=3Dde>Deutsche Telekom AG</SPAN> <BR><SPAN =
lang=3Den-gb>Supervisory=20
Board: <SPAN class=3D012202609-06032008>Prof. Dr Ulrich Lehner=20
</SPAN>(Chairman)</SPAN> <BR><SPAN lang=3Den-gb>Board of Management: =
Ren=E9 Obermann=20
(Chairman), </SPAN><BR><SPAN lang=3Den-gb>Dr. Karl-Gerhard Eick (Deputy=20
Chairman),</SPAN> <BR><SPAN lang=3Den-gb>Hamid Akhavan, Reinhard =
Clemens,=20
</SPAN><SPAN lang=3Dde>Timotheus H=F6ttges,&nbsp; Thomas =
Sattelberger</SPAN><SPAN=20
lang=3Den-gb></SPAN><SPAN lang=3Den-gb></SPAN> <BR><SPAN =
lang=3Dde>Commercial=20
register: Amtsgericht Bonn HRB 6794,</SPAN> <BR><SPAN =
lang=3Dde>corporate seat:=20
Bonn</SPAN> <BR><SPAN lang=3Dde>WEEEreg.no: =
DE50478376</SPAN></FONT></FONT> </P>
<DIV>&nbsp;</DIV></BODY></HTML>

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To: "'Sudhir Kumar Reddy'" <t.sudhirkumar@gmail.com>, <sip@ietf.org>
References: <52db68bc0806252333q5bb1a424g682e9e083a7b6b98@mail.gmail.com>
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Subject: Re: [Sip] can SDP contain 2 'M' lines
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> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of Sudhir Kumar Reddy
> Sent: Wednesday, June 25, 2008 11:34 PM
> To: sip@ietf.org
> Subject: [Sip] can SDP contain 2 'M' lines
> 
> Hi All,
> 
> In SDP can we have two 'M' lines for re-INVITE i.e. for call hold.
> 
> if yes what is RFC / draft

I have seen two responses that answer your first question and say 'yes'.
However, multiple m= lines are not used to put a call on hold.

-d

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From sip-bounces@ietf.org  Thu Jun 26 09:35:53 2008
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Subject: Re: [Sip] [Sipping] Progress on
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Hi folks,
=20
Just to get this queued since I've not yet seen a posting to SIP WG on
this topic, my attempt at bcc'ing SIP and keeping this discussion on
SIPPING only didn't work (I think I forgot to delete SIP in the too ;( =20
=20
But, please note that I would prefer (barring any disagreement from SIP
chairs) to keep this discussion on the SIPPING WG mailing list for now.
=20
Thanks,
Mary.=20

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<BODY>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D968043116-26062008>Hi folks,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D968043116-26062008></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D968043116-26062008>Just to get this queued since I've not yet =
seen a=20
posting to SIP WG on this topic, my attempt at bcc'ing SIP and keeping =
this=20
discussion on SIPPING only didn't work (I think I forgot to delete SIP =
in the=20
too ;(&nbsp; </SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D968043116-26062008></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D968043116-26062008>But, please note that I would prefer (barring =
any=20
disagreement from SIP chairs) to keep this discussion on the SIPPING WG =
mailing=20
list for now.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D968043116-26062008></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D968043116-26062008>Thanks,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D968043116-26062008>Mary. </SPAN></FONT></DIV></BODY></HTML>

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_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
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Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
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Subject: Re: [Sip] Toward the Evolution of SIP and Related Working Groups
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On Jun 26, 2008, at 7:36 AM, Paul Kyzivat wrote:

>
>
> Hannes Tschofenig wrote:
>
>> If you want to pick such as service provider then why does the  
>> chain of trust not work for you?  Hopefully you trust the provider  
>> you have chosen.
>
> Well, that seems obvious. If the SP is willing to trust *me*, then  
> it is probably willing to trust its other customers. If I am not  
> willing to trust its other customers, then I can't trust it.
>

I believe that my service provider has illegally collaborated with a  
rogue government organization to turn over call records in violation  
of federal law, and that when they got caught doing it, were able to  
lobby for special protections. They enjoy a legally-protected near  
monopoly when they ought to be up against the wall for treason, having  
conspired to violate the national constitution.

I might have to do business with the, but why the heck should I trust  
them?

--
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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(As SIP WG co chair)
=20
Please respond to this thread on the SIPPING list.
=20
As Mary has indicated, SIPPING have to agree the requirements and then
SIP can consider how best to develop the extensions.
=20
regards
=20
Keith


________________________________

	From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
Behalf Of Mary Barnes
	Sent: Thursday, June 26, 2008 5:33 PM
	To: sip@ietf.org
	Subject: Re: [Sip] [Sipping] Progress
ondraft-jesske-sipping-etsi-ngn-reason-03
=09
=09
	Hi folks,
	=20
	Just to get this queued since I've not yet seen a posting to SIP
WG on this topic, my attempt at bcc'ing SIP and keeping this discussion
on SIPPING only didn't work (I think I forgot to delete SIP in the too
;( =20
	=20
	But, please note that I would prefer (barring any disagreement
from SIP chairs) to keep this discussion on the SIPPING WG mailing list
for now.
	=20
	Thanks,
	Mary.=20


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
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<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.3243" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D000271817-26062008>(As SIP WG co chair)</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D000271817-26062008></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D000271817-26062008>Please respond to this thread on the SIPPING=20
list.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D000271817-26062008></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D000271817-26062008>As Mary has indicated, SIPPING have to agree =
the=20
requirements and then SIP can consider how best to develop the=20
extensions.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D000271817-26062008></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D000271817-26062008>regards</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D000271817-26062008></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
class=3D000271817-26062008>Keith</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> sip-bounces@ietf.org=20
  [mailto:sip-bounces@ietf.org] <B>On Behalf Of </B>Mary =
Barnes<BR><B>Sent:</B>=20
  Thursday, June 26, 2008 5:33 PM<BR><B>To:</B> =
sip@ietf.org<BR><B>Subject:</B>=20
  Re: [Sip] [Sipping] Progress=20
  ondraft-jesske-sipping-etsi-ngn-reason-03<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D968043116-26062008>Hi folks,</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D968043116-26062008></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D968043116-26062008>Just to get this queued since I've not yet =
seen a=20
  posting to SIP WG on this topic, my attempt at bcc'ing SIP and keeping =
this=20
  discussion on SIPPING only didn't work (I think I forgot to delete SIP =
in the=20
  too ;(&nbsp; </SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D968043116-26062008></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D968043116-26062008>But, please note that I would prefer =
(barring any=20
  disagreement from SIP chairs) to keep this discussion on the SIPPING =
WG=20
  mailing list for now.</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D968043116-26062008></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D968043116-26062008>Thanks,</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D968043116-26062008>Mary. =
</SPAN></FONT></DIV></BLOCKQUOTE></BODY></HTML>

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_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============1061096297==--


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To: Dean Willis <dean.willis@softarmor.com>
References: <E6C2E8958BA59A4FB960963D475F7AC30BE3C778B0@mail.acmepacket.com>	<485FEDBA.9090304@cisco.com>	<549C8D7E-8F0D-43C8-B6C3-B61D5D0DE238@softarmor.com>	<E6C2E8958BA59A4FB960963D475F7AC30BE7D5C137@mail.acmepacket.com>	<5744EABA-0861-4B95-BD91-2C4F0AC92BFE@softarmor.com>	<08d401c8d57a$dc7216d0$c2f0200a@cisco.com>	<4860AD84.60004@gmx.net>	<05d001c8d628$98c91510$c2f0200a@cisco.com>	<48614E77.9010904@alcatel-lucent.com>	<071b01c8d63f$2e39e910$c2f0200a@cisco.com>	<4861675E.1010507@alcatel-lucent.com>	<076c01c8d643$85c210a0$c2f0200a@cisco.com>	<0D5F89FAC29E2C41B98A6A762007F5D0D6E4AA@GBNTHT12009MSX.gb002.siemens.net>	<4862347B.8080405@cisco.com>	<0D5F89FAC29E2C41B98A6A762007F5D0D6E76C@GBNTHT12009MSX.gb002.siemens.net>	<C41BFCED3C088E40A8510B57B165C162313393@FIESEXC007.nsn-intra.net>	<0D5F89FAC29E2C41B98A6A762007F5D0D6E9B6@GBNTHT12009MSX.gb002.siemens.net>
	<4863528C.6040309@gmx.net> <48638D3E.4020908@cisco.com>
	<C9DC5874-E35D-4F04-86BC-D626DE160BBD@softarmor.com>
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Cc: sip@ietf.org, "Elwell, John" <john.elwell@siemens.com>,
	Dan Wing <dwing@cisco.com>
Subject: Re: [Sip] Toward the Evolution of SIP and Related Working Groups
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Amen!

	Paul

(That was legally protected religious speech - about as religious as you 
will ever get from me.)

Dean Willis wrote:
> 
> On Jun 26, 2008, at 7:36 AM, Paul Kyzivat wrote:
> 
>>
>>
>> Hannes Tschofenig wrote:
>>
>>> If you want to pick such as service provider then why does the chain 
>>> of trust not work for you?  Hopefully you trust the provider you have 
>>> chosen.
>>
>> Well, that seems obvious. If the SP is willing to trust *me*, then it 
>> is probably willing to trust its other customers. If I am not willing 
>> to trust its other customers, then I can't trust it.
>>
> 
> I believe that my service provider has illegally collaborated with a 
> rogue government organization to turn over call records in violation of 
> federal law, and that when they got caught doing it, were able to lobby 
> for special protections. They enjoy a legally-protected near monopoly 
> when they ought to be up against the wall for treason, having conspired 
> to violate the national constitution.
> 
> I might have to do business with the, but why the heck should I trust them?
> 
> -- 
> Dean
> 
> 
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From: "henry@sinnreich.net" <henry@sinnreich.net>
To: Paul Kyzivat <pkyzivat@cisco.com>, Dean Willis <dean.willis@softarmor.com>
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Trusting service providers?

>> I believe that my service provider has illegally collaborated with a
>> rogue government organization to turn over call records

Here is what happens elsewhere as well, in this instance without any
government involvement:

Germany fuming at Telekom's spy scandal
http://www.upi.com/Emerging_Threats/2008/05/27/Germany_fuming_at_Telekoms_sp
y_scandal/UPI-56191211919425/

Henry



On 6/26/08 11:23 AM, "Paul Kyzivat" <pkyzivat@cisco.com> wrote:

> Amen!
> 
> Paul
> 
> (That was legally protected religious speech - about as religious as you
> will ever get from me.)
> 
> Dean Willis wrote:
>> 
>> On Jun 26, 2008, at 7:36 AM, Paul Kyzivat wrote:
>> 
>>> 
>>> 
>>> Hannes Tschofenig wrote:
>>> 
>>>> If you want to pick such as service provider then why does the chain
>>>> of trust not work for you?  Hopefully you trust the provider you have
>>>> chosen.
>>> 
>>> Well, that seems obvious. If the SP is willing to trust *me*, then it
>>> is probably willing to trust its other customers. If I am not willing
>>> to trust its other customers, then I can't trust it.
>>> 
>> 
>> I believe that my service provider has illegally collaborated with a
>> rogue government organization to turn over call records in violation of
>> federal law, and that when they got caught doing it, were able to lobby
>> for special protections. They enjoy a legally-protected near monopoly
>> when they ought to be up against the wall for treason, having conspired
>> to violate the national constitution.
>> 
>> I might have to do business with the, but why the heck should I trust them?
>> 
>> -- 
>> Dean
>> 
>> 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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


From sip-bounces@ietf.org  Fri Jun 27 00:00:42 2008
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(As SIP WG cochair)

I would ask you to be aware that the current scheduling of the various
WG in Dublin is subject to considerable change.

Due to the need to schedule 3 BOFs and at least one WG getting their
request in late, things are going to have to move around a lot. You can
expect that the RAI sessions will be wall to wall, including Friday
morning.

Please make sure your travel arrangements accommodate this.

Regards

Keith
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From: Sigrid Thijs <sigrid.thijs@androme.com>
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To: rvmithra@gmail.com
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Subject: Re: [Sip] Client sending 'a=inactive' when put on hold
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Hello,

rvmithra@gmail.com wrote:
> Hi,
>  
>    Is there any Softphone client which sends media attributes: 'a= inactive'
>    when the call is put on hold?

The Intellivic Softphone uses 'inactive' to put a call on hold. You can 
download a demo version at http://demo.intellivic.com.

kind regards,

Sigrid

>    Tried SJ phone, Xlite, BT softphone and all of these send 'a=sendonly'.
>    The client should be capable of reg with SER. This is to test the msg
>    handling by a SBC, so SIPp can't be used as i need to check the RTP 
> flow as
>    well.
> 
>    Cheers,
>    Vishnu
>  
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
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From: "Jesske, R" <R.Jesske@telekom.de>
To: <mary.barnes@nortel.com>,
    <sipping@ietf.org>,
    <sip@ietf.org>
X-OriginalArrivalTime: 27 Jun 2008 08:29:42.0835 (UTC)
	FILETIME=[F2C88830:01C8D82F]
Subject: Re: [Sip] [Sipping] Progress on
	draft-jesske-sipping-etsi-ngn-reason-03
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Dear Mary,
Thank for your advice.
I think the most important is to say that the Reason header already =
exists in RFC3326. The draft draft-jesske-sipping-etsi-ngn-reason-03 =
clarifies the use of the Reason Header within Responses. because RFC =
3326 says:
=20
   Initially, the Reason header field defined here appears to be most

   useful for BYE and CANCEL requests, but it can appear in any request

   within a dialog, in any CANCEL request and in any response whose

   status code explicitly allows the presence of this header field.



   Note that the Reason header field is usually not needed in responses

   because the status code and the reason phrase already provide

   sufficient information.


This section is the problem we see and this needs to be clarified.
=20
With regard to the TISPAN requirements document, meanwhile the most are =
fulfilled and many of the services are deployed.
=20
With regard to the use of the Reason Header I will stripp of the main =
Requirement wich fits. This is the interoperability between the =
PSTN/ISDN network and SIP networks. So the TISPAN requirements document =
says:
=20
REQ-GEN-1:  =20

   All simulation services must provide interoperability with the =20

   PSTN/ISDN. By interoperability we mean that, in the case that a =20

   simulation or supplementary service is provided to one of the users =20

   when one of the endpoints is located in the PSTN and the other is =20

   located in the NGN IMS network, the user should receive the service =20

   without any degradation as if the service were provided in the native =
=20

   network. =20
=20
So the Reason header within responses will help to fulfil this there are =
certain cause values that are very specific for services. Like the =
example within the draft pointing to the Closed User Group service.
=20
So I will add some requirements to the draft for the certain cases =
described within the draft.
=20
Best Regards
=20
Roland
=20
=20
=20


  _____ =20

Von: sipping-bounces@ietf.org [mailto:sipping-bounces@ietf.org] Im =
Auftrag von Mary Barnes
Gesendet: Donnerstag, 26. Juni 2008 18:28
An: Jesske, Roland; sipping@ietf.org; sip@ietf.org
Betreff: Re: [Sipping] Progress on =
draft-jesske-sipping-etsi-ngn-reason-03


If the requirements in this document are agreed in SIPPING WG (the doc =
seems really sparse in terms of requirements at this point), THEN the =
work could progress to the SIP WG. But, we first need to agree =
requirements.=20
=20
I see your applicability section and earlier there's reference to the =
TISPAN requirements document (which is expired). Can you pull the =
specific requirements from that document into this one or at least let's =
discuss on the list?  In scanning that requirements document, it's not =
clear to me which requirement this header is intended to meet. If it is =
intended to support many of those simulation services, then that should =
be stated. And, so I believe this brings us to the question as to =
whether we are going to support those simulation services?=20
=20
Given that many of those services are being developed in BLISS, would it =
not be prudent to wait until we see the details of those services to =
suggest that we need this header?  Or has that already been discussed =
and I've missed it? If so, please provide us a pointer to that =
discussion.  In that case, we could agree that SIPPING WG does not need =
to spend much time on the requirements and we could more easily agree =
that this could move to SIP WG to be completed.  Or, are there other =
requirements for this header? =20
=20
And, just for clarification, are their plans to progress that other =
general requirements document or should the WG treat that document as we =
did in the past the 3GPP requirements document as a working document, =
from which to pull specific requirements and solution proposals?  My =
inclination would be the latter, but we need feedback from the WG =
(including ETSI/TISPAN/IMS folks, of course).=20
=20
And, just a note, I bcc'ed SIP WG on this thread, since I would prefer =
this thread stay only in SIPPING WG for now, in particular just to keep =
down the volume on the SIP list, which is already really busy.
=20
Thanks,
Mary.=20

  _____ =20

From: sipping-bounces@ietf.org [mailto:sipping-bounces@ietf.org] On =
Behalf Of Jesske, R
Sent: Thursday, June 26, 2008 10:57 AM
To: sipping@ietf.org; sip@ietf.org
Subject: [Sipping] Progress on draft-jesske-sipping-etsi-ngn-reason-03


Dear all,
at the moment I'm looking for comments on the =
http://tools.ietf.org/id/draft-jesske-sipping-etsi-ngn-reason-03.txt, to =
progress the work on that.=20
The reason in responses will be used within the IMS so progress of this =
draft is needed.
=20
Also I've got the comment that the work should be shifted to SIP WG.
=20
Thoughts? Comments?
=20
Best Regards
=20
Roland
=20
 <http://tools.ietf.org/id/draft-jesske-sipping-etsi-ngn-reason-03.txt>  =

=20

Deutsche Telekom AG=20
Zentrum Technik Einf=FChrung=20
Roland Jesske=20
Gateways, Protokolle, Konvergenz, TE32-1
Heinrich-Hertz-Str. 3-7,  64295 Darmstadt, =20
Postfach, 64307 Darmstadt (Postanschrift)=20
+496151 628 2766 (Tel)
+491718618445  (Mobile)=20
E-Mail: r.jesske@telekom.de=20

 <http://www.telekom.de/> http://www.telekom.de=20

Registerrechtliche Unternehmensangaben:=20
Deutsche Telekom AG
Aufsichtsrat: Prof. Dr Ulrich Lehner (Vorsitzender)
Vorstand: Ren=E9 Obermann (Vorsitzender),
Dr. Karl-Gerhard Eick (stellvertretender Vorsit-zender),
Hamid Akhavan, Reinhard Clemens, Timotheus H=F6ttges, Thomas =
Sattelberger=20
Handelsregister: Amtsgericht Bonn HRB 6794,=20
Sitz der Gesellschaft: Bonn=20
WEEE-Reg.-Nr.: DE50478376=20
***************************************************************=20
Deutsche Telekom AG=20
Supervisory Board: Prof. Dr Ulrich Lehner (Chairman)=20
Board of Management: Ren=E9 Obermann (Chairman),=20
Dr. Karl-Gerhard Eick (Deputy Chairman),=20
Hamid Akhavan, Reinhard Clemens, Timotheus H=F6ttges,  Thomas =
Sattelberger=20
Commercial register: Amtsgericht Bonn HRB 6794,=20
corporate seat: Bonn=20
WEEEreg.no: DE50478376=20

=20


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">


<META content=3D"MSHTML 6.00.2900.3268" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D394551408-27062008><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Dear Mary,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D394551408-27062008><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Thank for your advice.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D394551408-27062008><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>I think the most important is to say that the =
Reason header=20
already exists in RFC3326. The draft =
draft-jesske-sipping-etsi-ngn-reason-03=20
clarifies the use of the Reason Header within Responses. because RFC =
3326=20
says:</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D394551408-27062008><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D394551408-27062008><FONT =
face=3DArial=20
color=3D#0000ff size=3D2><PRE><FONT color=3D#000000 size=3D3>   =
Initially, the Reason header field defined here appears to be most
   useful for BYE and CANCEL requests, but it can appear in any request
   within a dialog, in any CANCEL request and in any response whose
   status code explicitly allows the presence of this header field.

   Note that the Reason header field is usually not needed in responses
   because the status code and the reason phrase already provide
   sufficient information.</FONT>
</PRE></FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D394551408-27062008><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>This section is the problem we see and this =
needs to be=20
clarified.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D394551408-27062008><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D394551408-27062008><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>With regard to the TISPAN requirements =
document, meanwhile=20
the most are fulfilled and many of the services are=20
deployed.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D394551408-27062008><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D394551408-27062008><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>With regard to the use of the Reason Header I =
will stripp=20
of the main Requirement wich fits. This is the interoperability between =
the=20
PSTN/ISDN network and SIP networks. So the TISPAN requirements document=20
says:</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D394551408-27062008><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN =
class=3D394551408-27062008><PRE>REQ-GEN-1:  =20
   All simulation services must provide interoperability with the =20
   PSTN/ISDN. By interoperability we mean that, in the case that a =20
   simulation or supplementary service is provided to one of the users =20
   when one of the endpoints is located in the PSTN and the other is =20
   located in the NGN IMS network, the user should receive the service =20
   without any degradation as if the service were provided in the native =
=20
   network.  </PRE></SPAN></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D394551408-27062008>So the=20
Reason header within responses will help to fulfil this there are =
certain cause=20
values that are very specific for services. Like the example within the =
draft=20
pointing to the Closed User Group service.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D394551408-27062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D394551408-27062008>So I=20
will add some requirements to the draft for the certain cases described =
within=20
the draft.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D394551408-27062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D394551408-27062008>Best=20
Regards</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D394551408-27062008></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D394551408-27062008>Roland</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D394551408-27062008></SPAN></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>
<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=3Dde dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>Von:</B> sipping-bounces@ietf.org=20
  [mailto:sipping-bounces@ietf.org] <B>Im Auftrag von </B>Mary=20
  Barnes<BR><B>Gesendet:</B> Donnerstag, 26. Juni 2008 =
18:28<BR><B>An:</B>=20
  Jesske, Roland; sipping@ietf.org; sip@ietf.org<BR><B>Betreff:</B> Re:=20
  [Sipping] Progress on=20
  draft-jesske-sipping-etsi-ngn-reason-03<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D625541316-26062008>If the requirements&nbsp;in this document =
are agreed=20
  in SIPPING WG (the doc seems really sparse in terms of requirements at =
this=20
  point), THEN the work could progress to the SIP WG. But, we first need =
to=20
  agree requirements. </SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D625541316-26062008></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D625541316-26062008>I see your applicability section and =
earlier there's=20
  reference to the TISPAN requirements document (which is expired). Can =
you pull=20
  the specific requirements from that document into this one or at least =
let's=20
  discuss on the list?&nbsp; In scanning that requirements document, =
it's not=20
  clear to me which requirement this header is intended to meet. If it =
is=20
  intended to support many of those simulation services, then that =
should be=20
  stated. And, so I believe this brings us to the question as to whether =
we are=20
  going to support those simulation services? </SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D625541316-26062008></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D625541316-26062008>Given that many of those services are being =
developed=20
  in BLISS, would it not be prudent to wait until we see the details of =
those=20
  services to suggest that we need this header?&nbsp; Or has that =
already been=20
  discussed and I've missed it? If so, please provide us a pointer to =
that=20
  discussion.&nbsp; In that case, we could agree that SIPPING WG does =
not need=20
  to spend much time on the requirements and we could more easily agree =
that=20
  this could move to SIP WG to be completed.&nbsp; Or, are there other=20
  requirements for this header?&nbsp; </SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D625541316-26062008></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D625541316-26062008>And, just for clarification, are their =
plans to=20
  progress that other general requirements document or should the WG =
treat that=20
  document as we did in the past the 3GPP requirements document as a =
working=20
  document, from which to pull specific requirements and solution=20
  proposals?&nbsp; My inclination would be the latter, but we need =
feedback from=20
  the WG (including ETSI/TISPAN/IMS folks, of course). =
</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D625541316-26062008></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D625541316-26062008>And, just a note, I bcc'ed SIP WG on this =
thread,=20
  since I would prefer this thread stay only in SIPPING WG for now, in=20
  particular just to keep down the volume on the SIP list, which is =
already=20
  really busy.</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D625541316-26062008></SPAN></FONT>&nbsp;</DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D625541316-26062008>Thanks,</SPAN></FONT></DIV>
  <DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff =
size=3D2><SPAN=20
  class=3D625541316-26062008>Mary. </SPAN></FONT></DIV><BR>
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> sipping-bounces@ietf.org=20
  [mailto:sipping-bounces@ietf.org] <B>On Behalf Of </B>Jesske,=20
  R<BR><B>Sent:</B> Thursday, June 26, 2008 10:57 AM<BR><B>To:</B>=20
  sipping@ietf.org; sip@ietf.org<BR><B>Subject:</B> [Sipping] Progress =
on=20
  draft-jesske-sipping-etsi-ngn-reason-03<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN class=3D043404315-26062008>Dear =

  all,</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN class=3D043404315-26062008>at =
the moment I'm=20
  looking for comments on the <A=20
  =
href=3D"http://tools.ietf.org/id/draft-jesske-sipping-etsi-ngn-reason-03.=
txt">http://tools.ietf.org/id/draft-jesske-sipping-etsi-ngn-reason-03.txt=
</A>,=20
  to progress the work on that.&nbsp;</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN class=3D043404315-26062008>The =
reason in=20
  responses will be used within the IMS so&nbsp;progress of this draft =
is=20
  needed.</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D043404315-26062008></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN class=3D043404315-26062008>Also =
I've got the=20
  comment that the work should be shifted to SIP WG.</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D043404315-26062008></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN =
class=3D043404315-26062008>Thoughts?=20
  Comments?</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D043404315-26062008></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN class=3D043404315-26062008>Best =

  Regards</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D043404315-26062008></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D043404315-26062008>Roland</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D043404315-26062008></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><A=20
  =
href=3D"http://tools.ietf.org/id/draft-jesske-sipping-etsi-ngn-reason-03.=
txt"></A></FONT>&nbsp;</DIV>
  <DIV>&nbsp;</DIV>
  <P align=3Dleft><SPAN lang=3Dde><FONT face=3DArial size=3D1>Deutsche =
Telekom=20
  AG</FONT></SPAN>&nbsp;<BR><SPAN lang=3Dde><FONT face=3DArial =
size=3D1>Zentrum=20
  Technik Einf=FChrung</FONT></SPAN> <BR><SPAN lang=3Dde><FONT =
face=3DArial=20
  size=3D1>Roland Jesske</FONT></SPAN> <BR><SPAN lang=3Dde><FONT =
face=3DArial=20
  size=3D1>Gateways, Protokolle, Konvergenz, =
TE32-1</FONT></SPAN><BR><SPAN=20
  lang=3Dde><FONT size=3D1><FONT face=3DArial><SPAN =
lang=3Dde><B>Heinrich-Hertz-Str.=20
  3-7,</B>&nbsp;</SPAN> 64295 Darmstadt,&nbsp;</FONT></FONT></SPAN> =
<BR><SPAN=20
  lang=3Dde><FONT face=3DArial size=3D1>Postfach, 64307 Darmstadt=20
  (Postanschrift)</FONT></SPAN>&nbsp;<BR><FONT face=3DArial =
size=3D1>+496151 628=20
  2766 (Tel)</FONT><BR><SPAN lang=3Dfr><FONT face=3DArial =
size=3D1>+491718618445=20
  &nbsp;(Mobile)</FONT></SPAN><SPAN lang=3Dde></SPAN> <BR><SPAN =
lang=3Dfr><FONT=20
  face=3DArial size=3D1>E-Mail: <A=20
  href=3D"mailto:r.jesske@telekom.de">r.jesske@telekom.de</A> =
</FONT></SPAN></P>
  <P align=3Dleft><SPAN lang=3Dfr></SPAN><SPAN lang=3Dfr><A=20
  href=3D"http://www.telekom.de/"><FONT face=3DArial=20
  size=3D2>http://www.telekom.de</FONT></A><FONT face=3DArial =
color=3D#0000ff size=3D2>=20
  </FONT></SPAN><SPAN lang=3Dde></P></SPAN><SPAN lang=3Dde></SPAN><SPAN =
lang=3Dde>
  <P class=3Deinzug align=3Dleft></SPAN><FONT size=3D1><FONT =
face=3DArial><SPAN=20
  lang=3Dde><U>Registerrechtliche Unternehmensangaben:</U></SPAN>=20
  <BR></FONT></FONT><SPAN lang=3Dde><FONT size=3D1><FONT =
face=3DArial>Deutsche Telekom=20
  AG<BR>Aufsichtsrat:&nbsp;Prof. Dr Ulrich Lehner<SPAN=20
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Date: Fri, 27 Jun 2008 12:34:50 -0400
Thread-Topic: Re: [Sip] Progress draft-holmberg-sip-keep
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> >>    A typical example is emergency calls.
> >
> > I don't get this one. You don't need to send keep-alive packets to make
> > an outbound call (be it urgent or not).

You need to send keepalives to keep the signaling path open, so that the other side can send back requests during the duration of the call. (in fact, the PSAP could send an UPDATE every 30 seconds to force that in case there is no keepalive)

> Also frequent UDP keep-alives are a total non-starter for mobile handsets, due
> to excessive battery consumption. Yet mobile handsets are probably the most
> common way to do emergency calls nowadays, for obvious availability reasons.

Christer's talking about unregistered devices making an emergency call, and keeping that call alive and usable. (since at least in the US, we have requirements to allow mobile sets and landlines to be able to make emergency calls regardless of paid service subscription)
I assume battery usage for SIP requests during an emergency call is not an issue, since it's already sending RTP the whole time, no?

-hadriel
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References: <E6C2E8958BA59A4FB960963D475F7AC30BE3C778B0@mail.acmepacket.com><408DFED6-302C-4525-BCD5-1E43B68369BE@softarmor.com><485FEDBA.9090304@cisco.com><549C8D7E-8F0D-43C8-B6C3-B61D5D0DE238@softarmor.com><E6C2E8958BA59A4FB960963D475F7AC30BE7D5C137@mail.acmepacket.com><5744EABA-0861-4B95-BD91-2C4F0AC92BFE@softarmor.com><08d401c8d57a$dc7216d0$c2f0200a@cisco.com>	<4860AD84.60004@gmx.net><05d001c8d628$98c91510$c2f0200a@cisco.com><48614E77.9010904@alcatel-lucent.com><071b01c8d63f$2e39e910$c2f0200a@cisco.com><4861675E.1010507@alcatel-lucent.com><076c01c8d643$85c210a0$c2f0200a@cisco.com><0D5F89FAC29E2C41B98A6A762007F5D0D6E4AA@GBNTHT12009MSX.gb002.siemens.net><4862347B.8080405@cisco.com><0D5F89FAC29E2C41B98A6A762007F5D0D6E76C@GBNTHT12009MSX.gb002.siemens.net><C41BFCED3C088E40A8510B57B165C162313393@FIESEXC007.nsn-intra.net><0D5F89FAC29E2C41B98A6A762007F5D0D6E9B6@GBNTHT12009MSX.gb002.siemens.net><4863528C.6040309@gmx.net><48638D3E.4020908@cisco.com>
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Cc: sip@ietf.org, "'Elwell, John'" <john.elwell@siemens.com>
Subject: Re: [Sip] SIP Identity
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Agreed.

-d

> -----Original Message-----
> From: Hadriel Kaplan [mailto:HKaplan@acmepacket.com] 
> Sent: Friday, June 27, 2008 8:06 AM
> To: Tschofenig, Hannes (NSN - FI/Espoo); ext Paul Kyzivat; 
> Hannes Tschofenig
> Cc: sip@ietf.org; Elwell, John; Dan Wing
> Subject: RE: SIP Identity
> 
> 
> 
> > -----Original Message-----
> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of
> > Tschofenig, Hannes (NSN - FI/Espoo)
> >
> > I am trying to figure out where the problems are:
> >
> > Now, it would be interesting to hear whether these guys 
> have actually
> > ever turned it on and used it. What was their experience? Did they
> > indeed encounter problems with certain SBC configurations?
> 
> I am fairly confident in saying that there could not be an 
> SBC that wouldn't break 4474 signatures, since it's basically 
> inherent in the definition of an SBC to modify some of the 
> things that 4474 signs. (ie, if it's not modifying them, it's 
> not really acting in an SBC "mode" anymore)
> Except of course when the SBC is the signer/verifier.
> 
> 
> > Would more folks use it when some of the weaknesses on the 
> E.164 number
> > usage are documented?
> 
> I doubt it.  Plenty of weaknesses of RTP have been 
> "documented", which haven't driven SRTP's uptake.  It will 
> take actual real-world cases of abuse to drive it, IMHO.  But 
> by then it will be late.  First there's the problem of making 
> it work across already deployed infrastructure, which 4474 
> doesn't, and there can be no "flag day".  Given how long it 
> takes us to standardize things, get them into code/hardware, 
> and get people to upgrade, we're talking years.  That's why I 
> want to work on it now, just in case we need it in the next few years.
> 
> 
> > How would a change to the signature algorithm impact the situation?
> 
> By making it work across deployed infrastructure.  More and 
> more Enterprise-SP SIP Trunks are getting added every day, 
> and inter-SP peering is continually growing.  If an Identity 
> algorithm has a lot of false positives (signatures broken but 
> the request is really ok), it won't be much use and people 
> will ignore it.  No algorithm has a chance of use if legit calls fail.
> 
> -hadriel

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Cc: "sip@ietf.org" <sip@ietf.org>
Subject: Re: [Sip] SIP Identity WG name
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Another vote for INSIPID :)

Hadriel Kaplan wrote:
> I don't know if people care, but since email is free: it would be nice to decide on a name for this potential group, and RETIRE is kinda a downer.
> 
> Potential names:
> RETIRE - "REal Time Identity and Role Expression"
> SIGNED - "Sip Identity for Generic Name and E.164 Deployments"
> SIGN - "Sip Identity for Global Names"
> RISE - "Roles and Identities in SIP Environments"
> 
> Or if you prefer a tounge-in-cheek approach:
> RESIGN - "Roles and Establishment for SIP Identities of Generic Nature" (which would be pronounced either as the act of signing again as in "re-sign", or of retiring as in "resign")
> INSIPID - "INternet SIP IDentities" (which could be pronounced either as "in SIP-ID", or the word insipid)
> 
> So far we have 3 votes for INSIPID and 1 vote for SIGNED.
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP 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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I like SIGNED. I agree that RETIRE wasn't the right name.

My 2 cents,
Dan
-- 
Dan York, CISSP, Director of Emerging Communication Technology
Office of the CTO    Voxeo Corporation     dyork@voxeo.com
Phone: +1-407-455-5859  Skype: danyork  http://www.voxeo.com
Blogs: http://blogs.voxeo.com  http://www.disruptivetelephony.com

Bring your web applications to the phone.
Find out how at http://evolution.voxeo.com



-----Original Message-----
From: Paul Kyzivat <pkyzivat@cisco.com>

Date: Fri, 27 Jun 2008 14:07:49 
To:Hadriel Kaplan <HKaplan@acmepacket.com>
Cc:"sip@ietf.org" <sip@ietf.org>
Subject: Re: [Sip] SIP Identity WG name


Another vote for INSIPID :)

Hadriel Kaplan wrote:
> I don't know if people care, but since email is free: it would be nice to decide on a name for this potential group, and RETIRE is kinda a downer.
>
> Potential names:
> RETIRE - "REal Time Identity and Role Expression"
> SIGNED - "Sip Identity for Generic Name and E.164 Deployments"
> SIGN - "Sip Identity for Global Names"
> RISE - "Roles and Identities in SIP Environments"
>
> Or if you prefer a tounge-in-cheek approach:
> RESIGN - "Roles and Establishment for SIP Identities of Generic Nature" (which would be pronounced either as the act of signing again as in "re-sign", or of retiring as in "resign")
> INSIPID - "INternet SIP IDentities" (which could be pronounced either as "in SIP-ID", or the word insipid)
>
> So far we have 3 votes for INSIPID and 1 vote for SIGNED.
>
> _______________________________________________
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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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> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Hannes Tschofenig
> >>
> > [JRE] Well, what about enterprise via service provider to enterprise? As
> > an enterprise provider, I would like to be able to use SIP identity, but
> > a precondition is that it gets through that intermediate service
> > provider(s).
> >
> Why would like to pick a service provider in the first place?

Want the top 10 reasons? (in no particular order)
1) Because most of their calls still go to/from the PSTN.
2) Because most of their calls are still addressed to/from using E.164 - they have no idea it's going to end up at another SIP-enabled Enterprise.
3) Because voice means business/money for many Enterprises - if it doesn't work they want someone to go fix it, someone to yell at, someone to monitor it, etc.
4) Because the target destination Enterprise may not allow incoming calls from anyone but *their* SP.
5) Because their SP has defined qos guarantees for VoIP traffic, if it's SIP-routed through them. (in some ways one could argue SIP has accomplished what RSVP was meant to do but did not)
6) Because SIP has significant interop issues, and the Enterprise knows their stuff works with the SP.
7) Because the IT guys can justify it more easily to execs, and won't get fired if the SP screws up. (if there even *is* an IT dept.)
8) Because email has NOT been a shining example for how VoIP could work.
9) Because they have constrained-resource WAN links and want to use low-bandwidth codecs over that, but want to let the SP transcode it to whatever will work with the far end.
10) Because if it ain't broke, they don't want to fix it. (ie, if it works it works)


> Why are
> they deploying B2BUAs?

Who, the SP or the Enterprise?  The answer is usually "control" ultimately, in some form or other.


> If you really want todo that then pick one that does not damage your
> security. You are the customer and you decide.

Damage your security how, and according to who's definition of security?  You think end-end is somehow inherently secure.  But neither end trusts the other.  They trust the middles (specifically their respective next-middles) more than the other ends.  That's the rub.


> If you want to pick such as service provider then why does the chain of
> trust not work for you?  Hopefully you trust the provider you have chosen.

I trust my provider, but not their peers as much, and less their peers' peers.  If I am using myprovider.com and I get a call from them saying it's from bob@myprovider.com I would be confident that's true.  I have much less confidence when I get a call from my provider saying the caller is admin@paypal.com.

But this topic isn't really about me or us anyway, imho.  Most SP customers blindly trust their SP's, imo - IETF'ers are an exception not the rule.  It's about an SP wanting to live-up to the trust its customers put in it.  They'd like to know that when they get calls from one of their peer SPs, they can trust calls that didn't originate at that peer.  They essentially had that in the PSTN, because they could ferret out trouble-makers over time, there's a forced hierarchy of providers, and the cost of calls and linking into SS7 self-limited many issues.  Whether that holds true for SIP remains to be seen.  I'd like to be ready before we find otherwise.


> .... just trying to figure out what the typical deployment cases are and
> why we don't see any of this in XMPP.

If people only used SIP for free IM, with email-type user@domain FQDN URIs only, and had one dominant vendor implementation, I bet the deployment landscape would be quite similar to XMPP.  It's not the protocol that causes it. (Well, that's not totally true, but close enough)

-hadriel

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From: Henry Sinnreich <hsinnrei@adobe.com>
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> 1) Because most of their calls still go to/from the PSTN.

I wonder if this may still be true:
- All employees have cell phones
- The most active talkers use the Blackberry
- Hard working folks in cubes (like myself :-)) have a desktop phone but
consider it a nuisance and never use it. I prefer to IM so as not to make
noise and to have privacy.
This leaves very few types of workers using the PSTN, probably in call
centers and those in support functions.

Does anyone have a report with numbers on this?

> 3) Because voice means business/money for many Enterprises

I wonder about this too. Except for the marketing and support folks, talking
on the phone lowers productivity. Productive employees use multimedia
communications with document/desktop sharing, document routing, etc.

Using voice at work (at some cost to the enterprise) seems to me just a bad
habit or bad legacy, and is rather the exception, not the rule.

BTW: Don't you have a plain cell phone, iPhone or Blackberry?

Henry


On 6/27/08 11:33 AM, "Hadriel Kaplan" <HKaplan@acmepacket.com> wrote:

> 
> 
>> -----Original Message-----
>> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
>> Hannes Tschofenig
>>>> 
>>> [JRE] Well, what about enterprise via service provider to enterprise? As
>>> an enterprise provider, I would like to be able to use SIP identity, but
>>> a precondition is that it gets through that intermediate service
>>> provider(s).
>>> 
>> Why would like to pick a service provider in the first place?
> 
> Want the top 10 reasons? (in no particular order)
> 1) Because most of their calls still go to/from the PSTN.
> 2) Because most of their calls are still addressed to/from using E.164 - they
> have no idea it's going to end up at another SIP-enabled Enterprise.
> 3) Because voice means business/money for many Enterprises - if it doesn't
> work they want someone to go fix it, someone to yell at, someone to monitor
> it, etc.
> 4) Because the target destination Enterprise may not allow incoming calls from
> anyone but *their* SP.
> 5) Because their SP has defined qos guarantees for VoIP traffic, if it's
> SIP-routed through them. (in some ways one could argue SIP has accomplished
> what RSVP was meant to do but did not)
> 6) Because SIP has significant interop issues, and the Enterprise knows their
> stuff works with the SP.
> 7) Because the IT guys can justify it more easily to execs, and won't get
> fired if the SP screws up. (if there even *is* an IT dept.)
> 8) Because email has NOT been a shining example for how VoIP could work.
> 9) Because they have constrained-resource WAN links and want to use
> low-bandwidth codecs over that, but want to let the SP transcode it to
> whatever will work with the far end.
> 10) Because if it ain't broke, they don't want to fix it. (ie, if it works it
> works)
> 
> 
>> Why are
>> they deploying B2BUAs?
> 
> Who, the SP or the Enterprise?  The answer is usually "control" ultimately, in
> some form or other.
> 
> 
>> If you really want todo that then pick one that does not damage your
>> security. You are the customer and you decide.
> 
> Damage your security how, and according to who's definition of security?  You
> think end-end is somehow inherently secure.  But neither end trusts the other.
> They trust the middles (specifically their respective next-middles) more than
> the other ends.  That's the rub.
> 
> 
>> If you want to pick such as service provider then why does the chain of
>> trust not work for you?  Hopefully you trust the provider you have chosen.
> 
> I trust my provider, but not their peers as much, and less their peers' peers.
> If I am using myprovider.com and I get a call from them saying it's from
> bob@myprovider.com I would be confident that's true.  I have much less
> confidence when I get a call from my provider saying the caller is
> admin@paypal.com.
> 
> But this topic isn't really about me or us anyway, imho.  Most SP customers
> blindly trust their SP's, imo - IETF'ers are an exception not the rule.  It's
> about an SP wanting to live-up to the trust its customers put in it.  They'd
> like to know that when they get calls from one of their peer SPs, they can
> trust calls that didn't originate at that peer.  They essentially had that in
> the PSTN, because they could ferret out trouble-makers over time, there's a
> forced hierarchy of providers, and the cost of calls and linking into SS7
> self-limited many issues.  Whether that holds true for SIP remains to be seen.
> I'd like to be ready before we find otherwise.
> 
> 
>> .... just trying to figure out what the typical deployment cases are and
>> why we don't see any of this in XMPP.
> 
> If people only used SIP for free IM, with email-type user@domain FQDN URIs
> only, and had one dominant vendor implementation, I bet the deployment
> landscape would be quite similar to XMPP.  It's not the protocol that causes
> it. (Well, that's not totally true, but close enough)
> 
> -hadriel
> 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

_______________________________________________
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From sip-bounces@ietf.org  Fri Jun 27 12:14:28 2008
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From: Dean Willis <dean.willis@softarmor.com>
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Apparently the SIP list's (and IETF list's) subscriber database got  
scrambled badly.

The admins restored a backup, but it was apparently a very old (like,  
2 year old) backup. So a lot of people that have unsubscribed in that  
period are now getting SIP mail, and a lot of people who subscribed  
(or changed addresses, like Robert Sparks) in that period have become  
unsubscribed.

The admins are working on it. I think that the list is currently  
"frozen", so it may be pointless to try and subscribe or unsubscribe  
at this time. It may be a day or two before the situation settles down.

I'm sorry. They're working on it. And above all (on the cover in large  
friendly letters) Don't Panic!

--
Dean

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From sip-bounces@ietf.org  Fri Jun 27 12:31:47 2008
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Henry Sinnreich <hsinnrei@adobe.com>, Hannes Tschofenig
	<Hannes.Tschofenig@gmx.net>, "Elwell, John" <john.elwell@siemens.com>
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> -----Original Message-----
> From: Henry Sinnreich [mailto:hsinnrei@adobe.com]
> Sent: Friday, June 27, 2008 3:11 PM
> To: Hadriel Kaplan; Hannes Tschofenig; Elwell, John
> Cc: sip@ietf.org; Paul Kyzivat; Dan Wing
> Subject: Re: [Sip] Toward the Evolution of SIP and Related Working Groups
>
> > 1) Because most of their calls still go to/from the PSTN.
>
> I wonder if this may still be true:
> - All employees have cell phones

All cell phones *are* on the PSTN, for all intents and purposes to which the email was about.

> > 3) Because voice means business/money for many Enterprises
>
> I wonder about this too. Except for the marketing and support folks,
> talking

And sales, and brick-and-mortar retailers, and food delivery, and repair shops, and hotels, and restaurants, and financial trading floors, and most things found in thick yellow books or on business cards.  I'm not saying commerce doesn't happen in other ways (duh), just that it also happens over voice.  I have no idea if it's more/less than email or http or whatever (and I bet the answer to that is regionally-specific), just that it's darn big.


> Using voice at work (at some cost to the enterprise) seems to me just a
> bad
> habit or bad legacy, and is rather the exception, not the rule.
>
> BTW: Don't you have a plain cell phone, iPhone or Blackberry?

See above - cell phones *are* the PSTN, at least so far they are.

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


From sip-bounces@ietf.org  Fri Jun 27 12:54:33 2008
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Date: Fri, 27 Jun 2008 12:54:17 -0700
From: Henry Sinnreich <hsinnrei@adobe.com>
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	Hannes Tschofenig <Hannes.Tschofenig@gmx.net>,
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> See above - cell phones *are* the PSTN, at least so far they are.

Any IMS folks here to agree with this?  :-)

Henry


On 6/27/08 12:31 PM, "Hadriel Kaplan" <HKaplan@acmepacket.com> wrote:

> 
> 
>> -----Original Message-----
>> From: Henry Sinnreich [mailto:hsinnrei@adobe.com]
>> Sent: Friday, June 27, 2008 3:11 PM
>> To: Hadriel Kaplan; Hannes Tschofenig; Elwell, John
>> Cc: sip@ietf.org; Paul Kyzivat; Dan Wing
>> Subject: Re: [Sip] Toward the Evolution of SIP and Related Working Groups
>> 
>>> 1) Because most of their calls still go to/from the PSTN.
>> 
>> I wonder if this may still be true:
>> - All employees have cell phones
> 
> All cell phones *are* on the PSTN, for all intents and purposes to which the
> email was about.
> 
>>> 3) Because voice means business/money for many Enterprises
>> 
>> I wonder about this too. Except for the marketing and support folks,
>> talking
> 
> And sales, and brick-and-mortar retailers, and food delivery, and repair
> shops, and hotels, and restaurants, and financial trading floors, and most
> things found in thick yellow books or on business cards.  I'm not saying
> commerce doesn't happen in other ways (duh), just that it also happens over
> voice.  I have no idea if it's more/less than email or http or whatever (and I
> bet the answer to that is regionally-specific), just that it's darn big.
> 
> 
>> Using voice at work (at some cost to the enterprise) seems to me just a
>> bad
>> habit or bad legacy, and is rather the exception, not the rule.
>> 
>> BTW: Don't you have a plain cell phone, iPhone or Blackberry?
> 
> See above - cell phones *are* the PSTN, at least so far they are.
> 
> -hadriel

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


From sip-bounces@ietf.org  Fri Jun 27 14:13:29 2008
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The SIP mailing list went insane last night and started running on a  
roughly 3-year old copy of its subscriber list.

The database problem has supposedly been corrected, but there may be  
subscribers who are not in the correct state.

If you want to receive SIP mailing list messages and aren't getting  
them, please re-subscribe.

If you want to NOT receive SIP mailing list messages and ARE getting  
them, please unsubscribe.

You can  manage subscriptions through the email.

By email, send a request to sip-request@ietf.org with either  
"subscribe" or "unsubscribe" (don't type the quotes)  in the body, as  
appropriate.

Supposedly, there's also a web interface for this, but I can't seem to  
find it at the moment.

Have a nice day.

--
Dean




_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From sip-bounces@ietf.org  Fri Jun 27 14:30:10 2008
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The Web interface for managing subscriptions to SIP was found for me  
by Vijay Gurbani:


https://www.ietf.org/mailman/listinfo/sip

Thanks, Vijay

--
Dean
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
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Cc: Dan Wing <dwing@cisco.com>
Subject: Re: [Sip] can SDP contain 2 'M' lines
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At 11:32 AM 6/26/2008, Dan Wing wrote:
>
>
> > -----Original Message-----
> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On
> > Behalf Of Sudhir Kumar Reddy
> > Sent: Wednesday, June 25, 2008 11:34 PM
> > To: sip@ietf.org
> > Subject: [Sip] can SDP contain 2 'M' lines
> >
> > Hi All,
> >
> > In SDP can we have two 'M' lines for re-INVITE i.e. for call hold.
> >
> > if yes what is RFC / draft
>
>I have seen two responses that answer your first question and say 'yes'.
>However, multiple m= lines are not used to put a call on hold.

indeed

1 (the same) m= line in sdp (offer) of reINVITE;
same Call-ID, From and To tag as original INVITE;
R-URI to same UA; (though I'm not sure if this R-URI can be to the 
Contact address in original 200 OK);
same c= line;
*different* (1) or *new* a= line from
         a=sendrecv
         to
         a=sendonly

the answer will be a=recvonly

(1) omitting a=sendrecv is ok in the original offer, since it is 
implied unless a direction tag is explicitly different in any offer.

see RFCs 3264 and 4566 for more details

James



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

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


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Subject: [Sip] [Sipping] Update: Outage on SIP Mailing List
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The Web interface for managing subscriptions to SIP was found for me  
by Vijay Gurbani:


https://www.ietf.org/mailman/listinfo/sip

Thanks, Vijay

--
Dean
_______________________________________________
Sipping mailing list  https://www.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://www.ietf.org/mailman/listinfo/sip
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Use sipping@ietf.org for new developments on the application of sip


From sip-bounces@ietf.org  Sat Jun 28 22:02:59 2008
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Subject: [Sip] Subscriber List Damage
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All -

I was asked by the IAOC to post a message to the IETF and SIP lists, to 
ensure that people were aware that the subscriber lists for the IETF and 
SIP lists were damaged as a result of an anomaly in TMDA and Mailman 
that occurred Thursday night.

Basically, TMDA misbehaved, and, in the process, caused Mailman to 
encounter a transient failure in the reading of its databases for these 
two lists.  As a result, rather than simply holding the mail and 
retrying it, Mailman decided to discard the current list databases and 
re-create them from 3-year-old data, for both the IETF and the SIP lists.

*sigh*

No email was lost to the system or the archives; however, some people 
may have missed some messages, or may still not be resubscribed to the list.

Of course we restored the files from backups; however, we want to make 
sure that everyone gets the mail they missed, and that everyone is 
subscribed to these lists who wishes to be subscribed.

So...

If you're reading this message in your email box, you're subscribed to 
the list identified in the subject line, and all should be okay.

If you're reading this message in the archives, wondering why you're not 
getting list mail, please take a moment to resubscribe yourself to the 
list, which should resolve your problem.

And regardless, if you feel you missed any mail, we do have the archives 
available for your reference.

IETF List Subscription Link:  https://www.ietf.org/mailman/listinfo/ietf
IETF List Archive Link:  http://www.ietf.org/mail-archive/web/ietf/

SIP List Subscription Link:  https://www.ietf.org/mailman/listinfo/sip
SIP List Archive Link:  http://www.ietf.org/mail-archive/web/sip/

We are in the home stretch of getting TMDA removed and replaced on the 
servers, and I apologize for any inconvenience caused by this issue. 
Because server problems apparently happen only in the dead of night, you 
can be sure that we feel any and all pain anyone may be experiencing.

If you need any assistance, please contact the IETF Secretariat, using 
the links at:  http://www.ietf.org/secretariat/

Thank you,
Glen Barney
IT Director
AMS (IETF Secretariat)
_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Thread-Topic: [Sip] Toward the Evolution of SIP and Related Working Groups
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References: <E6C2E8958BA59A4FB960963D475F7AC30EEDB2E3FB@mail.acmepacket.com>
	<C48A9379.612B%hsinnrei@adobe.com>
From: "DRAGE, Keith \(Keith\)" <drage@alcatel-lucent.com>
To: "Henry Sinnreich" <hsinnrei@adobe.com>,
	"Hadriel Kaplan" <HKaplan@acmepacket.com>,
	"Hannes Tschofenig" <Hannes.Tschofenig@gmx.net>,
	"Elwell, John" <john.elwell@siemens.com>
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Cc: sip@ietf.org, Paul Kyzivat <pkyzivat@cisco.com>, Dan Wing <dwing@cisco.com>
Subject: Re: [Sip] Toward the Evolution of SIP and Related Working Groups
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In the context of the argument Hadriel is putting forward, he is
correct.

IMS deployments exist, but you can still regard what is out there as
essentially IMS islands connected by the PSTN.

For example, I suspect any Skype call you make to an IMS user is via the
PSTN.

And standards for direct interconnection of the enterprise SIP networks
to IMS are relatively new, and we are still working on the bells and
whistles.

I would also note that some of the biggest users of mobiles for both
voice and messaging are under 18, and they don't have Blackberrys, and
the messaging they use is SMS over the CS network (i.e. PSTN), not over
IP, at least in the 3GPP side of the house.

Regards

Keith 

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On 
> Behalf Of Henry Sinnreich
> Sent: Friday, June 27, 2008 8:54 PM
> To: Hadriel Kaplan; Hannes Tschofenig; Elwell, John
> Cc: sip@ietf.org; Paul Kyzivat; Dan Wing
> Subject: Re: [Sip] Toward the Evolution of SIP and Related 
> Working Groups
> 
> > See above - cell phones *are* the PSTN, at least so far they are.
> 
> Any IMS folks here to agree with this?  :-)
> 
> Henry
> 
> 
> On 6/27/08 12:31 PM, "Hadriel Kaplan" <HKaplan@acmepacket.com> wrote:
> 
> > 
> > 
> >> -----Original Message-----
> >> From: Henry Sinnreich [mailto:hsinnrei@adobe.com]
> >> Sent: Friday, June 27, 2008 3:11 PM
> >> To: Hadriel Kaplan; Hannes Tschofenig; Elwell, John
> >> Cc: sip@ietf.org; Paul Kyzivat; Dan Wing
> >> Subject: Re: [Sip] Toward the Evolution of SIP and Related Working 
> >> Groups
> >> 
> >>> 1) Because most of their calls still go to/from the PSTN.
> >> 
> >> I wonder if this may still be true:
> >> - All employees have cell phones
> > 
> > All cell phones *are* on the PSTN, for all intents and purposes to 
> > which the email was about.
> > 
> >>> 3) Because voice means business/money for many Enterprises
> >> 
> >> I wonder about this too. Except for the marketing and 
> support folks, 
> >> talking
> > 
> > And sales, and brick-and-mortar retailers, and food delivery, and 
> > repair shops, and hotels, and restaurants, and financial trading 
> > floors, and most things found in thick yellow books or on business 
> > cards.  I'm not saying commerce doesn't happen in other ways (duh), 
> > just that it also happens over voice.  I have no idea if it's 
> > more/less than email or http or whatever (and I bet the 
> answer to that is regionally-specific), just that it's darn big.
> > 
> > 
> >> Using voice at work (at some cost to the enterprise) seems 
> to me just 
> >> a bad habit or bad legacy, and is rather the exception, 
> not the rule.
> >> 
> >> BTW: Don't you have a plain cell phone, iPhone or Blackberry?
> > 
> > See above - cell phones *are* the PSTN, at least so far they are.
> > 
> > -hadriel
> 
> _______________________________________________
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> This list is for NEW development of the core SIP Protocol Use 
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> 
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I'm still trying to clean up odds and ends from the recent meltdown of  
the mailing list. Sorry 'bout that.

--
Dean
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Thread-Topic: Proto writeup for draft-ietf-sip-rph-new-namespaces-03
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Subject: [Sip] Proto writeup for draft-ietf-sip-rph-new-namespaces-03
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(As SIP WG cochair)

I have just requested publication of
draft-ietf-sip-rph-new-namespaces-03

The Proto writeup for this document is as follows:

PROTO writeup for http://www.ietf.org/internet-drafts/
draft-ietf-sip-rph-new-
namespaces-03.txt: " IANA Registration of New Session Initiation
Protocol (SIP) Resource-
Priority Header Namespaces"

   (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-polk-sip-rph-new-namespaces-00 was submitted 26th February
2007 and expired 
26th August 2007.
-	draft-polk-sip-rph-new-namespaces-01 was submitted 9th July 2007
and expired 9th 
January 2008.
-	draft-ietf-sip-rph-new-namespaces-00 was submitted 16th October
2007 and expired 
16th April 2008.
-	draft-ietf-sip-rph-new-namespaces-01.txt was submitted 19th
November 2007 and 
expired 17th May 2008.
-	draft-ietf-sip-rph-new-namespaces-02.txt was submitted 21st
February 2008 and 
expires 21st August 2008.
-	draft-ietf-sip-rph-new-namespaces-03.txt was submitted 10th
March 2008 and 
expires 10th September 2008.

WGLC was initiated in the SIP WG on draft-ietf-sip-rph-new-namespaces-00
on 31st October 
2007 with comments requested by 14th November 2007.

Review was made and comments were received from: Janet Gunn, Joel
Halpern, John 
Rosenberg, Henning Schulzrinne, Hannes Tschofenig, Dean Willis. During
the course of the 
work comments have also been made by: Janet Gunn, Dale Worley.

The main reason for the document is to create the IANA namespaces, but
in the process of 
doing this it also requires one very small update to RFC 4412 in order
to create the 
OPTIONAL delimiter, which would be opaque to implementers otherwise, and
potentially 
causing implementations to not process the header (therefore message)
correctly.

One of the issues in the WGLC was the number of namespaces generated,
which apparently 
contravenes RFC 4412 which essentially says that one should reuse
existing namespaces if 
they are applicable, rather than going away and generating lots of new
ones. There is 
text in section 1 of the document justifying this large allocation.

   (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 has received review from a number of people whose interests
lie in this 
particular field, in addition to the normal WG responses.

   (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 shepherd has no concerns with this 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 addresses a limited part of the SIP community, but is of
significant 
interest in that part of the community. Within that limited community,
there is strong 
consensus behind the document.

   (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 
that the document is conformant with those guidelines.

For ID-NITS the checks against idnits 2.08.10 report no NITS found
except:

  == It seems as if not all pages are separated by form feeds - found 0
form
     feeds but 7 pages

which is a common fault which can be resolved readily during
publication.


   (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 contains only normative references. These references are
both standards 
track documents, are published and have been verified to be normative
references.

   (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 main purpose of this document is to create additional entries in an
existing IANA 
registry. The registry additions are clearly identified. The registry
requirements for 
this existing registry is standards track required, which this document
is.

   (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, and as
such there are no 
validation requirements.

   (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 document creates additional Session Initiation Protocol (SIP)
Resource-Priority 
namespaces to meet the requirements of the US Defense Information
Systems Agency, and 
places these namespaces in the IANA registry.

Working group summary.

There is consensus in the working group to publish this document. 

Document Quality

The document has received review by experts in the field as well as
members of the SIP 
working group.

Personnel

The document shepherd for this document was Keith Drage. The responsible
Area Director 
was Cullen Jennings. The IANA Expert(s) for the registries in this
document are <TO BE 
ADDED BY THE AD>.
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All (but particularly Jari and Adam) :

What does the body of a request mean if you get a SUBSCRIBE to the  
"xcap-diff" event, and that subscribe
contains a Require: recipient-list-subscribe ?

I think we may have created some ambiguity here. If I'm wrong and it's  
clear, where is it captured in the documents?

RjS
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From sip-bounces@ietf.org  Mon Jun 30 14:38:06 2008
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Thread-Topic: [Sip] Progress draft-holmberg-sip-keep
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From: "Kevin Johns" <K.Johns@CableLabs.com>
To: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>, <sip@ietf.org>
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Q1: I agree that the emergency services use case is important. However,
I would like to see the statement that some devices do not support
outbound but want to use the keep alive mechanism be further clarified
(what devices this applies to and why they would not implement outbound
- I think this has already been discussed on the list).

Q2: Based on my comments to Q1, I would like to see the requirements
narrowed in scope to only be in the context of a dialog initiating
request outside of a registration. The reason being that outbound is the
approach defined by the WG for client to Edge Proxy NAT traversal and I
don't think there is a strong need for more tools which allow the
creation of another NAT traversal solution. This narrowing of scope
would still allow support by devices which would not implement outbound
and are not providing edge proxy functionality.

Regards,
Kevin


-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
DRAGE, Keith (Keith)
Sent: Thursday, June 19, 2008 3:31 AM
To: sip@ietf.org
Cc: Christer Holmberg
Subject: [Sip] Progress draft-holmberg-sip-keep

(As SIP WG cochair)

We have been asked by the author of 

http://www.ietf.org/internet-drafts/draft-holmberg-sip-keep-01.txt

Whether the SIP WG can progress this document.

Because this draft arose as a result of the discussion of outbound, and
indeed seems to reuse the requirements from outbound, and these
requirements never really got handled in the SIPPING WG, it has been
agreed with the SIPPING chairs that we will handle this entirely within
SIP.

Now in order to ask for charter milestones, and indeed when we finally
present this to IESG, we will be asked for the level of support in the
WG, which is also predicated on does this fix a real problem, or is it
just a corner case with limited application. So:

QUESTION 1 TO SIP WG: Are the use cases sufficiently important to
proceed with this draft? The document states:

   Chapter 3.5 of draft-ietf-sip-outbound-13 [I-D.ietf-sip-outbound]
   defines two keep-alive techniques.  Even though the keep-alive
   techniques are separated from the Outbound mechanism
   [I-D.ietf-sip-outbound], it is currently not possible to indicate
   support of the keep-alive techniques without also indicating support
   for the Outbound mechanism.

   The Outbound mechanism is enabled during the UA registration phase.
   However, there are use-cases where the UA does not register itself,
   but still needs to be able to make calls and maintain NAT bindings
   open during the duration of that call.  A typical example is
   emergency calls.  There are also cases where entities do not support
   the Outbound mechanism, but still want to be able to indicate support
   and use the keep-alive techniques defined in [I-D.ietf-sip-outbound].

At first sight this is not the most inspiring declaration of the need
for the document. Please respond indicating whether you consider this a
useful draft, and propose text that you think would be useful in this
section. Conversely, if you think this draft is not useful and the WG
has other more important things to work on first, please also respond.

QUESTION 2 TO SIP WG: Do we have a robust set of requirements for
proceeding with this work? The document currently lists:

   REQ 1: It MUST be possible for a UA to indicate support of the keep-
   alive techniques defined [I-D.ietf-sip-outbound] if the UA supports
   only the keep-alive part of [I-D.ietf-sip-outbound].

   REQ 2: It MUST be possible for an edge proxy to indicate support of
   the keep-alive techniques defined [I-D.ietf-sip-outbound] if the edge
   poxy supports only the keep-alive part of [I-D.ietf-sip-outbound].

It would be desirable to agree these at the outset, and not revisit them
if we continue with the work. So if you require clarification,
modification, or addition to these two requirements, then please also
response with your questions and proposals.

I suggest we would like responses by 30th June 2008 in order to allow
the author to revise the document before the deadlines. Please note that
we are looking to make this decision on the list within this deadline
based on responses received, not leave it until the Dublin meeting.

Regards

Keith
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Cc: sip@ietf.org, IAOC <iaoc@ietf.org>, ietf@ietf.org
Subject: Re: [Sip] Subscriber List Damage
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1) Have you brought this up with the mailman folks? I've interacted with
    them and they seem like a responsive set of folks. I'm sure that 
this sort
    of thing would horrify them.
2) 3 years since the last backup? Oi.

       Mike

Glen wrote:
> All -
>
> I was asked by the IAOC to post a message to the IETF and SIP lists, 
> to ensure that people were aware that the subscriber lists for the 
> IETF and SIP lists were damaged as a result of an anomaly in TMDA and 
> Mailman that occurred Thursday night.
>
> Basically, TMDA misbehaved, and, in the process, caused Mailman to 
> encounter a transient failure in the reading of its databases for 
> these two lists.  As a result, rather than simply holding the mail and 
> retrying it, Mailman decided to discard the current list databases and 
> re-create them from 3-year-old data, for both the IETF and the SIP lists.
>
> *sigh*
>
> No email was lost to the system or the archives; however, some people 
> may have missed some messages, or may still not be resubscribed to the 
> list.
>
> Of course we restored the files from backups; however, we want to make 
> sure that everyone gets the mail they missed, and that everyone is 
> subscribed to these lists who wishes to be subscribed.
>
> So...
>
> If you're reading this message in your email box, you're subscribed to 
> the list identified in the subject line, and all should be okay.
>
> If you're reading this message in the archives, wondering why you're 
> not getting list mail, please take a moment to resubscribe yourself to 
> the list, which should resolve your problem.
>
> And regardless, if you feel you missed any mail, we do have the 
> archives available for your reference.
>
> IETF List Subscription Link:  https://www.ietf.org/mailman/listinfo/ietf
> IETF List Archive Link:  http://www.ietf.org/mail-archive/web/ietf/
>
> SIP List Subscription Link:  https://www.ietf.org/mailman/listinfo/sip
> SIP List Archive Link:  http://www.ietf.org/mail-archive/web/sip/
>
> We are in the home stretch of getting TMDA removed and replaced on the 
> servers, and I apologize for any inconvenience caused by this issue. 
> Because server problems apparently happen only in the dead of night, 
> you can be sure that we feel any and all pain anyone may be experiencing.
>
> If you need any assistance, please contact the IETF Secretariat, using 
> the links at:  http://www.ietf.org/secretariat/
>
> Thank you,
> Glen Barney
> IT Director
> AMS (IETF Secretariat)
> _______________________________________________
> Ietf mailing list
> Ietf@ietf.org
> https://www.ietf.org/mailman/listinfo/ietf

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Michael Thomas wrote:
> 1) Have you brought this up with the mailman folks? I've interacted with
>    them and they seem like a responsive set of folks. I'm sure that this 
> sort
>    of thing would horrify them.

I haven't yet, but will try.  They'll probably want more information 
than I can give them.

> 2) 3 years since the last backup? Oi.

No - not three years since the last backup.  I have MUCH more recent 
backups than that, of course!

Three years is the age of the file that I suspect mailman used to pull 
its old configuration from.  An old copy of the database for those two 
lists that was just sitting around, not being used.

Glen
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At Mon, 30 Jun 2008 15:48:10 -0700,
Michael Thomas wrote:
> 
> 1) Have you brought this up with the mailman folks? I've interacted with
>     them and they seem like a responsive set of folks. I'm sure that 
> this sort
>     of thing would horrify them.

I agree that this is horrifying.

More importantly, doesn't this mean that this is a problem we actually
need a solution for pronto? As I understand Glen's message, he's
saying that this is a bug in mailman triggered by some problem in
TMDA. I realize that TMDA is being replaced, but presumably Henrik's
code isn't perfect, so don't we have to worry about it triggering the
same behavior?

Glen, I'm sure there are some people on this list who understand
mailman well. I realize you may not have complete info, but if you can
provide us some more information--e.g., what file(s) got stomped and
which code you think stomped it--about what you think happened, maybe
they can help track it down?

-Ekr


> 2) 3 years since the last backup? Oi.
> 
>        Mike
> 
> Glen wrote:
> > All -
> >
> > I was asked by the IAOC to post a message to the IETF and SIP lists, 
> > to ensure that people were aware that the subscriber lists for the 
> > IETF and SIP lists were damaged as a result of an anomaly in TMDA and 
> > Mailman that occurred Thursday night.
> >
> > Basically, TMDA misbehaved, and, in the process, caused Mailman to 
> > encounter a transient failure in the reading of its databases for 
> > these two lists.  As a result, rather than simply holding the mail and 
> > retrying it, Mailman decided to discard the current list databases and 
> > re-create them from 3-year-old data, for both the IETF and the SIP lists.
> >
> > *sigh*
> >
> > No email was lost to the system or the archives; however, some people 
> > may have missed some messages, or may still not be resubscribed to the 
> > list.
> >
> > Of course we restored the files from backups; however, we want to make 
> > sure that everyone gets the mail they missed, and that everyone is 
> > subscribed to these lists who wishes to be subscribed.
> >
> > So...
> >
> > If you're reading this message in your email box, you're subscribed to 
> > the list identified in the subject line, and all should be okay.
> >
> > If you're reading this message in the archives, wondering why you're 
> > not getting list mail, please take a moment to resubscribe yourself to 
> > the list, which should resolve your problem.
> >
> > And regardless, if you feel you missed any mail, we do have the 
> > archives available for your reference.
> >
> > IETF List Subscription Link:  https://www.ietf.org/mailman/listinfo/ietf
> > IETF List Archive Link:  http://www.ietf.org/mail-archive/web/ietf/
> >
> > SIP List Subscription Link:  https://www.ietf.org/mailman/listinfo/sip
> > SIP List Archive Link:  http://www.ietf.org/mail-archive/web/sip/
> >
> > We are in the home stretch of getting TMDA removed and replaced on the 
> > servers, and I apologize for any inconvenience caused by this issue. 
> > Because server problems apparently happen only in the dead of night, 
> > you can be sure that we feel any and all pain anyone may be experiencing.
> >
> > If you need any assistance, please contact the IETF Secretariat, using 
> > the links at:  http://www.ietf.org/secretariat/
> >
> > Thank you,
> > Glen Barney
> > IT Director
> > AMS (IETF Secretariat)
> > _______________________________________________
> > Ietf mailing list
> > Ietf@ietf.org
> > https://www.ietf.org/mailman/listinfo/ietf
> 
> _______________________________________________
> Ietf mailing list
> Ietf@ietf.org
> https://www.ietf.org/mailman/listinfo/ietf
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


From sip-bounces@ietf.org  Mon Jun 30 19:29:59 2008
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From: Glen <glen@amsl.com>
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Unfortunately I really don't have anything but assumptions about what 
happened at this point.

An increase in TMDA activity caused the system to run out of resources. 
  We saw an extremely high load average and a sudden decrease in kernel 
buffer memory.  Processes started to fail to fork.  Our engineers 
connected in, and rebooted the server.

Upon reboot, we quickly discovered that the IETF list, which we 
moderate, was no longer letting us in with the normal list password.  A 
quick check showed that the database file config.pck was now only about 
50% of its original size, and that many subscribers had been removed - 
and others added.  Passwords and other settings had been "reverted" to 
pre-cutover values.  A comparison with our recent backup of the file 
showed massive differences - not just removals, but additions.  We 
hypothesize that mailman fell back to some type of cached copy of the 
database from 2005, which was also in the directory, and recreated the 
data from that.

Unfortunately, we have no way to verify what happened, and certainly 
don't want to try to cause it again.

I believe our solution is going to be to remove TMDA completely, which, 
I believe will eliminate the side-effects we've been seeing such as this.

I'm happy to discuss this further; however, I do not want to pollute the 
various lists with this.  I'm cc'ing the lists in this case so everyone 
understands we are responding to emails, but I'm going to take any 
further threads off-list to keep the lists focused on their primary 
purposes.

Glen

Eric Rescorla wrote:
> At Mon, 30 Jun 2008 15:48:10 -0700,
> Michael Thomas wrote:
>> 1) Have you brought this up with the mailman folks? I've interacted with
>>     them and they seem like a responsive set of folks. I'm sure that 
>> this sort
>>     of thing would horrify them.
> 
> I agree that this is horrifying.
> 
> More importantly, doesn't this mean that this is a problem we actually
> need a solution for pronto? As I understand Glen's message, he's
> saying that this is a bug in mailman triggered by some problem in
> TMDA. I realize that TMDA is being replaced, but presumably Henrik's
> code isn't perfect, so don't we have to worry about it triggering the
> same behavior?
> 
> Glen, I'm sure there are some people on this list who understand
> mailman well. I realize you may not have complete info, but if you can
> provide us some more information--e.g., what file(s) got stomped and
> which code you think stomped it--about what you think happened, maybe
> they can help track it down?
> 
> -Ekr
> 
> 
>> 2) 3 years since the last backup? Oi.
>>
>>        Mike
>>
>> Glen wrote:
>>> All -
>>>
>>> I was asked by the IAOC to post a message to the IETF and SIP lists, 
>>> to ensure that people were aware that the subscriber lists for the 
>>> IETF and SIP lists were damaged as a result of an anomaly in TMDA and 
>>> Mailman that occurred Thursday night.
>>>
>>> Basically, TMDA misbehaved, and, in the process, caused Mailman to 
>>> encounter a transient failure in the reading of its databases for 
>>> these two lists.  As a result, rather than simply holding the mail and 
>>> retrying it, Mailman decided to discard the current list databases and 
>>> re-create them from 3-year-old data, for both the IETF and the SIP lists.
>>>
>>> *sigh*
>>>
>>> No email was lost to the system or the archives; however, some people 
>>> may have missed some messages, or may still not be resubscribed to the 
>>> list.
>>>
>>> Of course we restored the files from backups; however, we want to make 
>>> sure that everyone gets the mail they missed, and that everyone is 
>>> subscribed to these lists who wishes to be subscribed.
>>>
>>> So...
>>>
>>> If you're reading this message in your email box, you're subscribed to 
>>> the list identified in the subject line, and all should be okay.
>>>
>>> If you're reading this message in the archives, wondering why you're 
>>> not getting list mail, please take a moment to resubscribe yourself to 
>>> the list, which should resolve your problem.
>>>
>>> And regardless, if you feel you missed any mail, we do have the 
>>> archives available for your reference.
>>>
>>> IETF List Subscription Link:  https://www.ietf.org/mailman/listinfo/ietf
>>> IETF List Archive Link:  http://www.ietf.org/mail-archive/web/ietf/
>>>
>>> SIP List Subscription Link:  https://www.ietf.org/mailman/listinfo/sip
>>> SIP List Archive Link:  http://www.ietf.org/mail-archive/web/sip/
>>>
>>> We are in the home stretch of getting TMDA removed and replaced on the 
>>> servers, and I apologize for any inconvenience caused by this issue. 
>>> Because server problems apparently happen only in the dead of night, 
>>> you can be sure that we feel any and all pain anyone may be experiencing.
>>>
>>> If you need any assistance, please contact the IETF Secretariat, using 
>>> the links at:  http://www.ietf.org/secretariat/
>>>
>>> Thank you,
>>> Glen Barney
>>> IT Director
>>> AMS (IETF Secretariat)
>>> _______________________________________________
>>> Ietf mailing list
>>> Ietf@ietf.org
>>> https://www.ietf.org/mailman/listinfo/ietf
>> _______________________________________________
>> Ietf mailing list
>> Ietf@ietf.org
>> https://www.ietf.org/mailman/listinfo/ietf
> 
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


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Thread-Topic: [Sip] SIP Identity WG name
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RETIRE sounds like a WG to discuss features that are going to be
deprecated.
and INSIPID sounds... insipid.


SIGNED for me unless...
Can we use SIPID?
is SIPID taken?
OK, it's not a real word but it means "what it says on the tin".






-----Original Message-----
From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
Dan York
Sent: 27 June 2008 19:14
To: Paul Kyzivat; sip-bounces@ietf.org; Hadriel Kaplan
Cc: sip@ietf.org
Subject: Re: [Sip] SIP Identity WG name

I like SIGNED. I agree that RETIRE wasn't the right name.

My 2 cents,
Dan
--
Dan York, CISSP, Director of Emerging Communication Technology
Office of the CTO    Voxeo Corporation     dyork@voxeo.com
Phone: +1-407-455-5859  Skype: danyork  http://www.voxeo.com
Blogs: http://blogs.voxeo.com  http://www.disruptivetelephony.com

Bring your web applications to the phone.
Find out how at http://evolution.voxeo.com



-----Original Message-----
From: Paul Kyzivat <pkyzivat@cisco.com>

Date: Fri, 27 Jun 2008 14:07:49
To:Hadriel Kaplan <HKaplan@acmepacket.com> Cc:"sip@ietf.org"
<sip@ietf.org>
Subject: Re: [Sip] SIP Identity WG name


Another vote for INSIPID :)

Hadriel Kaplan wrote:
> I don't know if people care, but since email is free: it would be nice
to decide on a name for this potential group, and RETIRE is kinda a
downer.
>
> Potential names:
> RETIRE - "REal Time Identity and Role Expression"
> SIGNED - "Sip Identity for Generic Name and E.164 Deployments"
> SIGN - "Sip Identity for Global Names"
> RISE - "Roles and Identities in SIP Environments"
>
> Or if you prefer a tounge-in-cheek approach:
> RESIGN - "Roles and Establishment for SIP Identities of Generic
Nature" (which would be pronounced either as the act of signing again as
in "re-sign", or of retiring as in "resign")
> INSIPID - "INternet SIP IDentities" (which could be pronounced either
as "in SIP-ID", or the word insipid)
>
> So far we have 3 votes for INSIPID and 1 vote for SIGNED.
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
_______________________________________________
Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


