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Date: Fri, 01 Feb 2013 17:36:32 -0500
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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To: "R.Jesske@telekom.de" <R.Jesske@telekom.de>,  Laura Liess <laura.liess.dt@googlemail.com>, "marianne.mohali@orange-ftgroup.com" <marianne.mohali@orange-ftgroup.com>,  Brett Tate <brett@broadsoft.com>, "Dale R. Worley" <worley@ariadne.com>
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Cc: SIPCORE <sipcore@ietf.org>
Subject: [sipcore] Verify draft-ietf-sipcore-rfc4244bis-callflows-02.txt
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A new version of 4244bis-callflows has been published, to address the 
comments we received in the WGLC review at the end of last year.

I've sent this *To* the people who provided comments.
I would appreciate your verifying that your comments were properly 
addressed. (Please comment back one way or the other.)

This is *not* a new WGLC. So please don't bring up *new* issues now. We 
are past that. I just want to verify that the comments already raised 
have been handled. We are holding the bis to publish concurrently with 
the callflows, so is the last thing holding them up.

	Thanks,
	Paul


-------- Original Message --------
Subject: [sipcore] I-D Action: 
draft-ietf-sipcore-rfc4244bis-callflows-02.txt
Date: Tue, 29 Jan 2013 12:49:12 -0800
From: internet-drafts@ietf.org
To: i-d-announce@ietf.org
CC: sipcore@ietf.org


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

	Title           : Session Initiation Protocol (SIP) History-Info Header 
Call Flow Examples
	Author(s)       : Mary Barnes
                           Francois Audet
                           Shida Schubert
                           Detecon International Gmbh
                           Christer Holmberg
	Filename        : draft-ietf-sipcore-rfc4244bis-callflows-02.txt
	Pages           : 46
	Date            : 2013-01-29

Abstract:
    This document describes use cases and documents call flows which
    require the History-Info header field to capture the Request-URIs as
    a Session Initiation Protocol (SIP) Request is retargeted.  The use
    cases are described along with the corresponding call flow diagrams
    and messaging details.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-sipcore-rfc4244bis-callflows

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-sipcore-rfc4244bis-callflows-02

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-sipcore-rfc4244bis-callflows-02


Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

_______________________________________________
sipcore mailing list
sipcore@ietf.org
https://www.ietf.org/mailman/listinfo/sipcore




From rjsparks@nostrum.com  Mon Feb  4 07:49:44 2013
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Date: Mon, 04 Feb 2013 09:49:39 -0600
From: Robert Sparks <rjsparks@nostrum.com>
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Subject: [sipcore] Registration for SIPit30 extended to Feb 10
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We are set for a great SIPit, but we still have room for more folks. The 
host is extending the registration deadline through Feb 10.

RjS

On 1/16/13 1:19 PM, Robert Sparks wrote:
> The registration deadline for SIPit 30 is February 4, just over two 
> weeks away.
> If you plan to attend, but have not yet registered, please do so today.
>
> Note that we will be testing against RTCWeb implementations during 
> Tuesdays multiparty test.
> See www.sipit.net for more info.
>
> RjS
>
> On 11/26/12 9:34 AM, Robert Sparks wrote:
>> Registration for SIPit 30 is open at http://www.regonline.com/sipit30
>>
>> SIPit 30 will be hosted by Cisco in Raleigh-Durham, North Carolina, 
>> February 18-22 2013.
>>
>> Additional details are available at http://www.sipit.net.
>>
>> See you there!
>>
>> RjS
>>
>


From worley@shell01.TheWorld.com  Mon Feb  4 12:53:41 2013
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From: worley@ariadne.com (Dale R. Worley)
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Subject: Re: [sipcore] Verify draft-ietf-sipcore-rfc4244bis-callflows-02.txt
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My previous comments have been addressed acceptably.  But there are a
few new nits that I've spotted:

** General

* On page 19

   Alice       example.com     Gold          Silver       Agent

   |              |              |             |            |
   | INVITE F1    |              |             |            |
   |------------->|              |             |            |
   |              |              |             |            |
   |              |  INVITE F2   |             |            |
   |              |------------->|             |            |
   |              |              |             |            |
   |              |  302 Moved Temporarily F3  |            |
   |              |<-------------|             |            |
   |              |              |             |            |
   |             ACK             |             |            |
   |---------------------------->|             |            |
   |              |              |             |            |

The ACK should be shown thus:

   |              |              |             |            |
   |              |  ACK         |             |            |
   |              |------------->|             |            |
   |              |              |             |            |

* On page 27, the top ACK should be shown thus:

   |              |              |             |            |
   |              |  ACK         |             |            |
   |              |------------->|             |            |
   |              |              |             |            |

* On page 31, the top ACK should be shown thus:

   |              |              |             |            |
   |              |  ACK         |             |            |
   |              |------------->|             |            |
   |              |              |             |            |

* On page 40, there are two occurrences of:

   To: <sip:sip:tgruu.7hs==jd7vnzga5w7fajsc7-ajd6fabz0f8g5@example.com\
    ;gr>

"sip:sip:" should be replaced with "sip:".

** http://www.ietf.org/mail-archive/web/sipcore/current/msg05489.html

* "5. In draft-ietf-sipcore-rfc4244bis-callflows-01, examples 3.6 and 3.7
do not show the History-Info value in the final 200 response to the
UAC.  This final value would be useful, in particular, because they
give further examples of how proxies process History-Info."

In neither of these examples is the final 200 response shown.  I
assume that this is the authors' intention.

** http://www.ietf.org/mail-archive/web/sipcore/current/msg05429.html

* "But in draft-ietf-sipcore-rfc4244bis-callflows-01, Reason header
values for 1xx responses are not generated.  See message F8 in section
3.1."

rfc4244bis-callflows-02 does not show Reason headers with 1xx values.
I expect this is because the authors intend for this to be how
History-Info operates.  But that does mean we are committed to
updating rfc4244bis to state that (because it currently says that 1xx
responses set Reason headers).

Dale

From brett@broadsoft.com  Sat Feb  9 11:49:31 2013
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From: Brett Tate <brett@broadsoft.com>
To: SIPCORE <sipcore@ietf.org>
Thread-Topic: Verify draft-ietf-sipcore-rfc4244bis-callflows-02.txt
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Howdy,

Most of my comments have been adequately addressed.

Thanks,
Brett

-----

Concerning my various comments to include Max-Forwards, the values are good=
 enough for me; however I was surprised to see some of the decrementing and=
 initial values.  For instance, section 3.1 F5 and F13 contain initial valu=
es within ACK reflecting the related INVITE instead of 70.  And for instanc=
e, section 3.1 F6 and F9 reflect the proxy decrementing the value again for=
 INVITE sent to new location because of 302 or timeout.

If not intentional, the figure labels are of the format "Figure 1: Figure 1=
." and not consistently located (i.e. before or after detailed messages).

Section 3.2 F3: top Via received should be on the second via.

Section 3.4 picture: 302 ACK arrow reflection wrong source.  The same appli=
es to sections 3.6 and 3.7.

Concerning http://www.ietf.org/mail-archive/web/sipcore/current/msg05265.ht=
ml comment 2, section 3.5 was not modified; thus I'll assume that the trans=
port switch from UDP to TCP was intentional (i.e. registered UDP Contact bu=
t INVITE sent over TCP).

Section 3.9 F2: Added Max-Forwards to response.

New comment (if want to address): figure 1 indicates a "retransmit INVITE" =
with arrow after having received a 180.


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From: <marianne.mohali@orange.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "R.Jesske@telekom.de" <R.Jesske@telekom.de>, Laura Liess <laura.liess.dt@googlemail.com>, "Brett Tate" <brett@broadsoft.com>, "Dale R. Worley" <worley@ariadne.com>
Thread-Topic: [sipcore] Verify draft-ietf-sipcore-rfc4244bis-callflows-02.txt
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Date: Thu, 14 Feb 2013 13:11:36 +0000
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Verify draft-ietf-sipcore-rfc4244bis-callflows-02.txt
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Hello,

Most of my comments have been perfectly addressed but I have still some con=
cerns with the following changes:

MM#1 =A73.6 in F3: there is an "mp" tag in a Contact header; I guess it's a=
 mistake and it should be a History-Info header (as in F4).

MM#2 =A73.6 in F6: The History-Info header is
History-Info: <sip:bob@example.com>;index=3D1 			[MM] ok [/MM]
History-Info: <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302>;\
      		   index=3D1.1;rc=3D1 				[MM] ok [/MM]
History-Info: <sip:carol@example.com>;index=3D1.2;mp=3D1 	[MM] ok [/MM]
History-Info: <sip:carol@192.0.2.4?Reason=3DSIP%3Bcause%3D480>;\
                     index=3D1.2.1;rc=3D1.2 				[MM] nok, it is a cause URI=
 parameter that should be set to "480" (call deflection) following RFC4458.=
 The Reason header is escaped in the SIP URI only when it is a real receive=
d respond (it is the case with the 302) [/MM]
History-Info: <sip:vm@example.com;\
                     target=3Dsip:bob%40example.com>;\
                     index=3D1.3;mp=3D1 				[MM] nok, The hi-entry with the=
 voicemail should be a copy of the R-URI so with the cause=3D480 URI parame=
ter. Additionally, the vm is finally reached because Carol does not reply, =
so the no reply reason (cause=3D408) should also be reflected in a cause UR=
I parameter following RFC4458. Hence, the hi-entry should be History-Info: =
<sip:vm@example.com;target=3Dsip:bob%40example.com;cause=3D480;cause=3D408>=
;                    index=3D1.3;mp=3D1  =3D> Oops, I think we have a probl=
em... This is due to the fact that at the same time we have successive forw=
arding (1 after the other) with Carol at the second-to-last position and we=
 try to join B's vm (1st diverting user). IMO, we should add some text stat=
ing that depending on which voicemail is reached (Bob or Carol), the call f=
orwarding reason (Bob or Carol's diverting reason) in the cause URI paramet=
er should be set accordingly (to be consistent). Here the target vm is Bob =
so only cause=3D480 should be used and Carol's reason for not answering the=
 call is lost. [/MM]
[MM]If someone have a better/other idea, please don't hesitate... [/MM]
History-Info: <sip:vm@192.0.2.6;\
                     target=3Dsip:bob%40example.com>;\
                     index=3D1.3.1;rc=3D1.3 				[MM] nok, same comment

But IMO it should be:

History-Info: <sip:bob@example.com>;index=3D1=20
History-Info: <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302>;\=20
                     index=3D1.1;rc=3D1=20=20=20=20=20=20=20=20=20
History-Info: <sip:carol@example.com>;index=3D1.2;mp=3D1=20
History-Info: <sip:carol@192.0.2.4;cause=3D480>;\
                     index=3D1.2.1;rc=3D1.2=20
History-Info: <sip:vm@example.com;\
                     target=3Dsip:bob%40example.com;cause=3D480>;\
                     index=3D1.3;mp=3D1
History-Info: <sip:vm@192.0.2.6;\
                     target=3Dsip:bob%40example.com;cause=3D480>;\
                     index=3D1.3.1;rc=3D1.3

MM#3 =A73.7 in F4 and followings:=20
In the hi-entry the reason-text has been added.=20
I have 2 comments:=20
1. why adding the reason-text? (and why not ;)=3D> ok for me=20
2. if the reason-text is added, it should be "escaped" in the same way as t=
he cause parameter.=20=20
So that, the hi-entry:
History-Info: <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302\
                         ;text=3D"Moved Temporarily">\
                 ;index=3D1.1;rc=3D1

Should be:
History-Info: <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302\
                         %3Btext%3D"Moved Temporarily">\
                 ;index=3D1.1;rc=3D1

Here is my feedback.

Best Regards,
Marianne

-----Message d'origine-----
De=A0: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] De la par=
t de Paul Kyzivat
Envoy=E9=A0: vendredi 1 f=E9vrier 2013 23:37
=C0=A0: R.Jesske@telekom.de; Laura Liess; MOHALI Marianne OLNC/OLN; Brett T=
ate; Dale R. Worley
Cc=A0: SIPCORE
Objet=A0: [sipcore] Verify draft-ietf-sipcore-rfc4244bis-callflows-02.txt

A new version of 4244bis-callflows has been published, to address the comme=
nts we received in the WGLC review at the end of last year.

I've sent this *To* the people who provided comments.
I would appreciate your verifying that your comments were properly addresse=
d. (Please comment back one way or the other.)

This is *not* a new WGLC. So please don't bring up *new* issues now. We are=
 past that. I just want to verify that the comments already raised have bee=
n handled. We are holding the bis to publish concurrently with the callflow=
s, so is the last thing holding them up.

	Thanks,
	Paul


-------- Original Message --------
Subject: [sipcore] I-D Action:=20
draft-ietf-sipcore-rfc4244bis-callflows-02.txt
Date: Tue, 29 Jan 2013 12:49:12 -0800
From: internet-drafts@ietf.org
To: i-d-announce@ietf.org
CC: sipcore@ietf.org


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

	Title           : Session Initiation Protocol (SIP) History-Info Header=20
Call Flow Examples
	Author(s)       : Mary Barnes
                           Francois Audet
                           Shida Schubert
                           Detecon International Gmbh
                           Christer Holmberg
	Filename        : draft-ietf-sipcore-rfc4244bis-callflows-02.txt
	Pages           : 46
	Date            : 2013-01-29

Abstract:
    This document describes use cases and documents call flows which
    require the History-Info header field to capture the Request-URIs as
    a Session Initiation Protocol (SIP) Request is retargeted.  The use
    cases are described along with the corresponding call flow diagrams
    and messaging details.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-sipcore-rfc4244bis-callflows

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-sipcore-rfc4244bis-callflows-02

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-sipcore-rfc4244bis-callflows-=
02


Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

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From: Ofer Goren <oferg@avaya.com>
To: "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: question about B2B and Forking
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--_000_D9AD3D1627F52341B829F32E6DDE089149E014C0rvilmail2010_
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Hi All.
I know B2B behavior is not defined with too much details in the relevant SI=
P RFCs. I do ask for your opinion:
In forking scenarios, if B2B forks an INVITE to several UAS-es, should it f=
orward downstream all the provisional responses to the generating UAC? Or, =
should it forward only one," absorbing" the rest? Should it even wait for p=
rovisional responses, or should it generates its own toward the UAC, indepe=
ndent to the upstream side of the session?
If it should forward all provisional responses downstream, it in fact creat=
es several early dialogs for which it should also handle transactions such =
as PRACK (in case of reliable provisional responses), or UPDATEs and such.

So what do you think a B2B should do?

Thanks!
Ofer


_______________________________________________________________
If you lend someone $20, and never see that person again, it was probably w=
orth the investment.



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<body lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div class=3D"WordSection1">
<p class=3D"MsoNormal">Hi All.<o:p></o:p></p>
<p class=3D"MsoNormal">I know B2B behavior is not defined with too much det=
ails in the relevant SIP RFCs. I do ask for your opinion:<o:p></o:p></p>
<p class=3D"MsoNormal">In forking scenarios, if B2B forks an INVITE to seve=
ral UAS-es, should it forward downstream all the provisional responses to t=
he generating UAC? Or, should it forward only one,&#8221; absorbing&#8221; =
the rest? Should it even wait for provisional
 responses, or should it generates its own toward the UAC, independent to t=
he upstream side of the session?<o:p></o:p></p>
<p class=3D"MsoNormal">If it should forward all provisional responses downs=
tream, it in fact creates several early dialogs for which it should also ha=
ndle transactions such as PRACK (in case of reliable provisional responses)=
, or UPDATEs and such.
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">So what do you think a B2B should do?<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Thanks!<o:p></o:p></p>
<p class=3D"MsoNormal">Ofer<o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Century Gothic&quot=
;,&quot;sans-serif&quot;;color:#6D6E71;background:white"><o:p>&nbsp;</o:p><=
/span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Century Gothic&quot=
;,&quot;sans-serif&quot;;color:#6D6E71;background:white"><o:p>&nbsp;</o:p><=
/span></p>
<p class=3D"MsoNormal"><span style=3D"color:#C00000">______________________=
_________________________________________</span><span style=3D"color:#C0000=
0"><o:p></o:p></span></p>
<p class=3D"MsoNormal"><i><span style=3D"font-size:10.0pt;font-family:&quot=
;Verdana&quot;,&quot;sans-serif&quot;;color:#707070">If you lend someone $2=
0, and never see that person again, it was probably worth the investment.
</span></i><i><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;=
,&quot;sans-serif&quot;;color:gray"><o:p></o:p></span></i></p>
<p class=3D"MsoNormal"><i><span style=3D"font-size:10.0pt;font-family:&quot=
;Arial&quot;,&quot;sans-serif&quot;;color:gray">&nbsp;<o:p></o:p></span></i=
></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</body>
</html>

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From: Brett Tate <brett@broadsoft.com>
To: Ofer Goren <oferg@avaya.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: question about B2B and Forking
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Hi Ofer,

The appropriate behavior depends upon the functionality that the B2BUA is p=
roviding.

Within normal call originations with devices compliant to RFC 3261, RFC 326=
2, and RFC 3311, it is typically best to act more like a proxy.

However since there are still devices that don't support RFC 3262, RFC 3311=
, or forking proxies, the B2BUA might have to act differently for interoper=
ability reasons.  Similarly, the B2BUA may need to act differently for serv=
ice reasons.


From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf =
Of Ofer Goren
Sent: Sunday, February 17, 2013 11:23 AM
To: sipcore@ietf.org
Subject: [sipcore] question about B2B and Forking

Hi All.
I know B2B behavior is not defined with too much details in the relevant SI=
P RFCs. I do ask for your opinion:
In forking scenarios, if B2B forks an INVITE to several UAS-es, should it f=
orward downstream all the provisional responses to the generating UAC? Or, =
should it forward only one," absorbing" the rest? Should it even wait for p=
rovisional responses, or should it generates its own toward the UAC, indepe=
ndent to the upstream side of the session?
If it should forward all provisional responses downstream, it in fact creat=
es several early dialogs for which it should also handle transactions such =
as PRACK (in case of reliable provisional responses), or UPDATEs and such.

So what do you think a B2B should do?

Thanks!
Ofer


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<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Hi Ofer,<o:p></o:p></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">The appropriate behavi=
or depends upon the functionality that the B2BUA is providing.<o:p></o:p></=
span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Within normal call ori=
ginations with devices compliant to RFC 3261, RFC 3262, and RFC 3311, it is=
 typically best to act more like a proxy.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">However since there ar=
e still devices that don&#8217;t support RFC 3262, RFC 3311, or forking pro=
xies, the B2BUA might have to act differently for interoperability reasons.=
&nbsp; Similarly, the B2BUA may need to act differently
 for service reasons.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> sipcore-=
bounces@ietf.org [mailto:sipcore-bounces@ietf.org]
<b>On Behalf Of </b>Ofer Goren<br>
<b>Sent:</b> Sunday, February 17, 2013 11:23 AM<br>
<b>To:</b> sipcore@ietf.org<br>
<b>Subject:</b> [sipcore] question about B2B and Forking<o:p></o:p></span><=
/p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Hi All.<o:p></o:p></p>
<p class=3D"MsoNormal">I know B2B behavior is not defined with too much det=
ails in the relevant SIP RFCs. I do ask for your opinion:<o:p></o:p></p>
<p class=3D"MsoNormal">In forking scenarios, if B2B forks an INVITE to seve=
ral UAS-es, should it forward downstream all the provisional responses to t=
he generating UAC? Or, should it forward only one,&#8221; absorbing&#8221; =
the rest? Should it even wait for provisional
 responses, or should it generates its own toward the UAC, independent to t=
he upstream side of the session?<o:p></o:p></p>
<p class=3D"MsoNormal">If it should forward all provisional responses downs=
tream, it in fact creates several early dialogs for which it should also ha=
ndle transactions such as PRACK (in case of reliable provisional responses)=
, or UPDATEs and such.
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">So what do you think a B2B should do?<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Thanks!<o:p></o:p></p>
<p class=3D"MsoNormal">Ofer<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</body>
</html>

--_000_576A8B541C219D4E9CEB1DF8C19C7B8819697E23MBX06citservers_--

From internet-drafts@ietf.org  Sun Feb 17 17:26:03 2013
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Subject: [sipcore] I-D Action: draft-ietf-sipcore-sip-websocket-07.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Session Initiation Protocol Core Working =
Group of the IETF.

	Title           : The WebSocket Protocol as a Transport for the Session In=
itiation Protocol (SIP)
	Author(s)       : Inaki Baz Castillo
                          Jose Luis Millan Villegas
                          Victor Pascual
	Filename        : draft-ietf-sipcore-sip-websocket-07.txt
	Pages           : 21
	Date            : 2013-02-17

Abstract:
   The WebSocket protocol enables two-way realtime communication between
   clients and servers in web-based applications.  This document
   specifies a WebSocket sub-protocol as a reliable transport mechanism
   between SIP (Session Initiation Protocol) entities to enable usage of
   SIP in web-oriented deployments.  This document normatively updates
   RFC 3261.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-sipcore-sip-websocket

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-sipcore-sip-websocket-07

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-sipcore-sip-websocket-07


Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


From ibc@aliax.net  Sun Feb 17 17:31:49 2013
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 18 Feb 2013 02:31:28 +0100
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Subject: Re: [sipcore] I-D Action: draft-ietf-sipcore-sip-websocket-07.txt
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Hi all,

First of all I'm really sorry for the delay in submitting this new
revision 07 of draft-ietf-sipcore-sip-websocket.

This new revision of the document includes the following changes based
on feedback received in the SIPCORE WG maillist:

- Removed mention to RFC 4168 "SIP STCP" in section "SIP URI Transport
Parameter" (thanks to Martin Vopatek).

- Section 6 "Connection Keep Alive" improved and normative statements
(RFC 2119) removed (thanks to Robert Sparks).

- Section 7 "Authentication" fixed and improved (thanks to Robert Sparks).

- Section 9.1 "Secure WebSocket Connection" (under "Security
Considerations") now mentions that SIP TLS certificate checks
specified in RFC 3261 or RFC 5922 don't apply when using SIP over
WebSocket transport (and RFC 5922 "Domain Certificates in SIP" added
to "Informative References").

- "Abstract" and "Introduction" improved by removing "new" keyword
(not good for RFCs) (thanks to Robert Sparks).

- ABNF in 5.2.1 and 5.2.2 (Via Transport Parameter and SIP URI
Transport Parameter) improved by using =3D/ syntax (thanks to Robert
Sparks and Paul Kyzivat).

- RFC 2617 "HTTP Authentication: Basic and Digest Access
Authentication" removed from "References" since it is no longer
mentioned in the document. Instead the document points to the RFC 3261
section describing Digest usage in SIP protocol.


Thanks a lot to all and best regards.




2013/2/18  <internet-drafts@ietf.org>:
>
> A New Internet-Draft is available from the on-line Internet-Drafts direct=
ories.
>  This draft is a work item of the Session Initiation Protocol Core Workin=
g Group of the IETF.
>
>         Title           : The WebSocket Protocol as a Transport for the S=
ession Initiation Protocol (SIP)
>         Author(s)       : Inaki Baz Castillo
>                           Jose Luis Millan Villegas
>                           Victor Pascual
>         Filename        : draft-ietf-sipcore-sip-websocket-07.txt
>         Pages           : 21
>         Date            : 2013-02-17
>
> Abstract:
>    The WebSocket protocol enables two-way realtime communication between
>    clients and servers in web-based applications.  This document
>    specifies a WebSocket sub-protocol as a reliable transport mechanism
>    between SIP (Session Initiation Protocol) entities to enable usage of
>    SIP in web-oriented deployments.  This document normatively updates
>    RFC 3261.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-sipcore-sip-websocket
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-sipcore-sip-websocket-07
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-sipcore-sip-websocket-07
>
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore



--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 18 Feb 2013 02:31:31 +0100
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Hi Robert, I've submited a new revision 07 of
draft-ietf-sipcote-sip-websocket, some comments inline:



2012/11/27 Robert Sparks <rjsparks@nostrum.com>:
> AD Review: draft-ietf-sipcore-sip-websocket-06.txt
>
> Summary: There are a few issues to address with a revised ID before
> progressing to IETF LC
>
> Overall, this is a very well written, easy to read draft. Thank you for
> making it this clear.
> There are a couple of small problems to correct, and a few issues to
> consider before moving
> forward:
>
> * Section 6 starts with "_This section is non-normative_", but then conta=
ins
>   normative statements. RECOMMENDED is the same as SHOULD. This is a
> stronger
>   requirement for Ping than RFC 6455 places on websockets. If it was the
> intent
>   to restate what RFC 6455 required, the language should be "RFC 6455 say=
s
> to...".
>   If it was the intent to make a stronger requirement, then this section
> _is_
>   normative.
>
>   To avoid the appearance of placing a new normative requirement on the
> implementation
>   of keep-alives, I suggest replacing the next to last paragraph of this
> section with
>   "The indication and use of the CRLF NAT keep-alive mechanism defined fo=
r
> SIP
>    connection-oriented transports in RFC5626 section 3.5.1 or RFC6223 are=
,
> of course,
>    usable over the transport defined in this specification."

Done in rev 07.



> * Section 7 starts with "_This section is non-normative_" but then
>   contain normative statements.
>
>   Section 7 could be rewritten to not use 2119, but that would take more
> than
>   changing the words to lower case. The gist is 'the SIP and WebSocket sp=
ecs
>   tell you these things.'
>
>   It also currently confuses SIP Digest with HTTP Digest, and incorrectly
> suggests
>   that SIP Digest is a SHOULD strength requirement in 3261 (it is a MUST
> there).
>
>   It would be better to say "This section describes how authentication is
>   acheived through the requirements in RFCs 6445, 6265, and 3261."
>   The MAY in paragraph 3 can be changed to "is allowed to".
>   The last paragraph can say
>    ...", authentication can be requested at the SIP protocol level. Note
>     that RFC3261 requires that all SIP implementations (which includes
>     implementations of this specification) implement Digest Authorization
>     (see RFC3261, Section 26.3.1)."

Done in rev 07.



> * The document should more explicitly call out that none of SIP TLS
> certificate
>   checks specified in RFC3261 or RFC5922 will be made when using this
> transport.
>   Instead, only the checks specified by RFC6455 will be made. The
> certificates
>   that are appropriate for SIP over TLS over TCP will probably not be
> appropriate
>   for SIP over secure websockets (it would only be coincidence if the sam=
e
> cert
>   worked - the two checks will look in different places in the certificat=
e).

Done in rev 07 within the "Secure WebSocket Connection" subsection
under "Security Considerations" section.



> * Should there be some additional discussion of what an element should do=
 if
>   someone clicks on "sip:foo.example.com;transport=3Dws" or receives that=
 in a
>   contact in a 302 or in a Refer-To: header field?  To: header field?
>   Taken to an extreme, what's a proxy that understands sip over websocket=
s
>   supposed to do if it gets a message containing Route header fields (thi=
nk
>   preloaded routes) where the next route value is for a transport=3Dws ho=
p
> that
>   a connection doesn't yet exist for?

Typicall WebSocket capable SIP proxies/servers won't be able to
establish outgoing WS connections and, ever less, typical SIP WS
clients (browsers) won't be able to receive incoming WS connections.
Thus a SIP proxy receiving a request with a ;ws preloaded Route would
fail due to "unsupported transport" when performing RFC 3263 rules.

However I expect that SIP WebSocket clients should/will implement
Outbound and GRUU and work in enviroments in which proxies and
registrars also implement Outbound and GRUU. Therefore the problem you
mention would not occur since both Outbound and GRUU avoid it. The SIP
WebSocket UA would provide a GRUU based Contact when establishing
dialogs and thus the remote peer would send a GRUU URI in the Refer-To
header.



> * I'm ok proceeding with the IANA Considerations sections as they current=
ly
>   stand, but I wonder if instead of what's in 10.3 and 10.4, this documen=
t
>   should create a new registry for the transport values. That way someone
>   could search the registry for the strings "WS" or "WSS" and find someth=
ing
>   meaningful.

I'm ok with what you suggest, but I don't think I can start such a new
registry. If you consider that the document requires a new registry
please let me know how to help with it.



> Nits:
>
>  - RFCs never change. Years from now, this won't be "new". Please conside=
r
>    removing "new" from the abstract and the Introduction. The abstract wo=
uld
>    be stronger if you replaced "enable usage of SIP in new scenarios" wit=
h
>    "enable usage of SIP in web-oriented deployments."

Done in rev 07.



>  - In the introduction, you say "a [new] reliable and message boundary
> transport".
>    Did you mean "a reliable, and message-boundary preserving, transport" =
?

Done in rev 07.




>  - At the end of 5.2.3, s/regardless the Via/regardless of whether the Vi=
a/

Fixed in rev 07.



>  - In A.1, to further avoid this section appearing to be normative, I
> suggest
>    changing "Therefore the SIP WebSocket Client constructs" to "A SIP
> WebSocket
>    client running in such an environment can construct".

Done in rev 07.



>  - In section 6, what is the value of the sentence "A future WebSocket
> protocol
>    extension providing a similar keep alive mechanism could also be used"=
?
> This
>    specification says the same thing if the sentence is removed, and I
> suggest
>    removing it.

You are right. Done in rev 07.



>  - (no change suggested - just pointing something out for future work)
>    The ABNF in 5.2.1 could have used =3D/ and not have restated the other
> values.
>    It would have been nice if 5.2.2 could have as well, but the way
> transport-param
>    is defined in RFC3261 prevents it. It's probably better to keep the fo=
rm
> of 5.2.1
>    and 5.2.2 the same.

I've rewritten it as Paul and you suggest:

  transport =3D/ "WS" / "WSS"

  transport-param =3D/ "transport=3D" "ws"



Thanks a lot for you useful feedback and improvements.

Best regards.



--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From worley@shell01.TheWorld.com  Mon Feb 18 13:06:29 2013
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From: worley@ariadne.com (Dale R. Worley)
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Cc: sipcore@ietf.org, R.Jesske@telekom.de, laura.liess.dt@googlemail.com
Subject: Re: [sipcore] Verify draft-ietf-sipcore-rfc4244bis-callflows-02.txt
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> From: <marianne.mohali@orange.com>
> Subject: RE: [sipcore] Verify draft-ietf-sipcore-rfc4244bis-callflows-02.txt
> 
> MM#1 B'3.6 in F3: there is an "mp" tag in a Contact header; I guess
> it's a mistake and it should be a History-Info header (as in F4).

No, that's to show that the contact <sip:carol@example.com> was
generated by "mapping" from the URI with index 1,
<sip:bob@example.com>.  This is described in 4244bis section 8.
(There are a number of such Contact headers in the callflows.)

> MM#2 B'3.6 in F6: The History-Info header is
> History-Info: <sip:bob@example.com>;index=1 			[MM] ok [/MM]
> History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302>;\
> 		   index=1.1;rc=1 				[MM] ok [/MM]
> History-Info: <sip:carol@example.com>;index=1.2;mp=1 	[MM] ok [/MM]
> History-Info: <sip:carol@192.0.2.4?Reason=SIP%3Bcause%3D480>;\
> 		     index=1.2.1;rc=1.2 				
> [MM] nok, it is a cause URI parameter that should be set to "480"
> (call deflection) following RFC4458. The Reason header is escaped in
> the SIP URI only when it is a real received respond (it is the case
> with the 302) [/MM]

I believe the cause is supposed to be 408 (Timeout) rather than 480,
and it is correct for the proxy to synthesize a 408 cause here because
of 4244bis section 10.2 paragraph 2.  (And many SIP systems treat
timing out while waiting for a response as if they received an
explicit 408 response.)

This mistake of 480 for 408 is also present in the request-line of F6
and the History-Info of F7.

(Confusion can easily arise because 480 (Temporarily Unavailable) is
also a plausible cause for diversion to VM.)

> History-Info: <sip:vm@example.com;\
> 		     target=sip:bob%40example.com>;\
> 		     index=1.3;mp=1 				
> [MM] nok, The hi-entry with the voicemail should be a copy of the
> R-URI so with the cause=480 URI parameter. Additionally, the vm is
> finally reached because Carol does not reply, so the no reply reason
> (cause=408) should also be reflected in a cause URI parameter
> following RFC4458. Hence, the hi-entry should be History-Info:
> <sip:vm@example.com;target=sip:bob%40example.com;cause=480;cause=408>;
> index=1.3;mp=1  => Oops, I think we have a problem... This is due to
> the fact that at the same time we have successive forwarding (1 after
> the other) with Carol at the second-to-last position and we try to
> join B's vm (1st diverting user). IMO, we should add some text stating
> that depending on which voicemail is reached (Bob or Carol), the call
> forwarding reason (Bob or Carol's diverting reason) in the cause URI
> parameter should be set accordingly (to be consistent). Here the
> target vm is Bob so only cause=480 should be used and Carol's reason
> for not answering the call is lost. [/MM]
> [MM]If someone have a better/other idea, please don't hesitate... [/MM]

Beware that message F6 does not correspond to entry 1.3 but rather to
entry 1.3.1, which is derived from 1.3 by internal redirection.  (It
might be worth adding a note in the text to warn the reader!)  So it
is entry 1.3.1 that must have the cause parameter, to match the
request-URI.

However, since the 1.3.1 request-URI has a cause parameter, it is
likely that 1.3 request-URI has the same cause parameter.

The messier problem is not really present.  There is only one cause at
this point, the implicit 408 response from Carol.  And there may be
complexities concerning what the diversion URI should be, but that is
the proxy's problem, not a problem in designing 4244bis, and we do not
have to solve it here, we only have to show the History-Info that
comes from this particular proxy's decisions.

> History-Info: <sip:vm@192.0.2.6;\
> 		     target=sip:bob%40example.com>;\
> 		     index=1.3.1;rc=1.3
> [MM] nok, same comment
> 
> But IMO it should be:
> 
> History-Info: <sip:bob@example.com>;index=1 
> History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302>;\ 
> 		     index=1.1;rc=1         
> History-Info: <sip:carol@example.com>;index=1.2;mp=1 
> History-Info: <sip:carol@192.0.2.4;cause=480>;\
> 		     index=1.2.1;rc=1.2 
> History-Info: <sip:vm@example.com;\
> 		     target=sip:bob%40example.com;cause=480>;\
> 		     index=1.3;mp=1
> History-Info: <sip:vm@192.0.2.6;\
> 		     target=sip:bob%40example.com;cause=480>;\
> 		     index=1.3.1;rc=1.3
> 
> MM#3 B'3.7 in F4 and followings: 
> In the hi-entry the reason-text has been added. 
> I have 2 comments: 
> 1. why adding the reason-text? (and why not ;)=> ok for me 
> 2. if the reason-text is added, it should be "escaped" in the same way
> as the cause parameter.  
> So that, the hi-entry:
> History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302\
> 			 ;text="Moved Temporarily">\
> 		 ;index=1.1;rc=1
> 
> Should be:
> History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302\
> 			 %3Btext%3D"Moved Temporarily">\
> 		 ;index=1.1;rc=1

The header field as it stands is:

   History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302\
                              ;text="Moved Temporarily">\
                 ;index=1.1;rc=1

but the escaping is incorrect.  The Reason header would be:

    Reason: SIP;cause=302;text="Moved Temporarily"

so the 'hname' is 'Reason' and the 'hvalue' is
'SIP;cause=302;text="Moved Temporarily"', so the URI should be

<sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302%3Btext%3D%34Moved%20Temporarily%34>

There are 4 instances of this problem, in 3.7 F4, F5, F6, and F7

Dale

From pkyzivat@alum.mit.edu  Mon Feb 18 15:10:21 2013
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Subject: Re: [sipcore] AD Review: draft-ietf-sipcore-sip-websocket-06.txt
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On 2/17/13 8:31 PM, IÃ±aki Baz Castillo wrote:

>> * I'm ok proceeding with the IANA Considerations sections as they currently
>>    stand, but I wonder if instead of what's in 10.3 and 10.4, this document
>>    should create a new registry for the transport values. That way someone
>>    could search the registry for the strings "WS" or "WSS" and find something
>>    meaningful.
>
> I'm ok with what you suggest, but I don't think I can start such a new
> registry. If you consider that the document requires a new registry
> please let me know how to help with it.

We just went through this by establishing a new registry for the 
priority header field.

But in that case, the draft that provoked the creation of the new 
registry was in a different WG, and we thought it weird for that draft 
to establish a new registry for a sip header field.

In this case everything is within sipcore, so I don't see any reason why 
we couldn't do everything within this draft.

Robert, what do you think?

	Paul

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From: Ofer Goren <oferg@avaya.com>
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Thread-Topic: question about B2B and Forking
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Date: Tue, 19 Feb 2013 13:02:23 +0000
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Subject: [sipcore] question about B2B and Forking
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Hi All.
I know B2B behavior is not defined with too much details in the relevant SI=
P RFCs. I do ask for your opinion:
In forking scenarios, if B2B forks an INVITE to several UAS-es, should it f=
orward downstream all the provisional responses to the generating UAC? Or, =
should it forward only one," absorbing" the rest? Should it even wait for p=
rovisional responses, or should it generates its own toward the UAC, indepe=
ndent to the upstream side of the session?
If it should forward all provisional responses downstream, it in fact creat=
es several early dialogs for which it should also handle transactions such =
as PRACK (in case of reliable provisional responses), or UPDATEs and such.

So what do you think a B2B should do?

Thanks!
Ofer


_______________________________________________________________
If you lend someone $20, and never see that person again, it was probably w=
orth the investment.



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<div class=3D"WordSection1">
<p class=3D"MsoNormal">Hi All.<o:p></o:p></p>
<p class=3D"MsoNormal">I know B2B behavior is not defined with too much det=
ails in the relevant SIP RFCs. I do ask for your opinion:<o:p></o:p></p>
<p class=3D"MsoNormal">In forking scenarios, if B2B forks an INVITE to seve=
ral UAS-es, should it forward downstream all the provisional responses to t=
he generating UAC? Or, should it forward only one,&#8221; absorbing&#8221; =
the rest? Should it even wait for provisional
 responses, or should it generates its own toward the UAC, independent to t=
he upstream side of the session?<o:p></o:p></p>
<p class=3D"MsoNormal">If it should forward all provisional responses downs=
tream, it in fact creates several early dialogs for which it should also ha=
ndle transactions such as PRACK (in case of reliable provisional responses)=
, or UPDATEs and such.
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">So what do you think a B2B should do?<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Thanks!<o:p></o:p></p>
<p class=3D"MsoNormal">Ofer<o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Century Gothic&quot=
;,&quot;sans-serif&quot;;color:#6D6E71;background:white"><o:p>&nbsp;</o:p><=
/span></p>
<p class=3D"MsoNormal"><span style=3D"font-family:&quot;Century Gothic&quot=
;,&quot;sans-serif&quot;;color:#6D6E71;background:white"><o:p>&nbsp;</o:p><=
/span></p>
<p class=3D"MsoNormal"><span style=3D"color:#C00000">______________________=
_________________________________________<o:p></o:p></span></p>
<p class=3D"MsoNormal"><i><span style=3D"font-size:10.0pt;font-family:&quot=
;Verdana&quot;,&quot;sans-serif&quot;;color:#707070">If you lend someone $2=
0, and never see that person again, it was probably worth the investment.
</span></i><i><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;=
,&quot;sans-serif&quot;;color:gray"><o:p></o:p></span></i></p>
<p class=3D"MsoNormal"><i><span style=3D"font-size:10.0pt;font-family:&quot=
;Arial&quot;,&quot;sans-serif&quot;;color:gray">&nbsp;<o:p></o:p></span></i=
></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</body>
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From radhika.r.roy.civ@mail.mil  Tue Feb 19 07:17:59 2013
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From: "Roy, Radhika R CIV USARMY (US)" <radhika.r.roy.civ@mail.mil>
To: Ofer Goren <oferg@avaya.com>
Thread-Topic: question about B2B and Forking (UNCLASSIFIED)
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Date: Tue, 19 Feb 2013 15:17:46 +0000
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Subject: Re: [sipcore] question about B2B and Forking (UNCLASSIFIED)
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Classification: UNCLASSIFIED
Caveats: NONE

Ofer:

In this case it not B2BUA spec per se. It will be rather specific to
"Forking Specifications using B2BUA." If we want to have a interoperable
implementation of forking using B2BUA. Is it really needed?

BR/Radhika


-----Original Message-----
From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf
Of Ofer Goren
Sent: Tuesday, February 19, 2013 8:02 AM
To: sipcore@ietf.org
Subject: [sipcore] question about B2B and Forking

Hi All.

I know B2B behavior is not defined with too much details in the relevant SIP
RFCs. I do ask for your opinion:

In forking scenarios, if B2B forks an INVITE to several UAS-es, should it
forward downstream all the provisional responses to the generating UAC? Or,
should it forward only one," absorbing" the rest? Should it even wait for
provisional responses, or should it generates its own toward the UAC,
independent to the upstream side of the session?

If it should forward all provisional responses downstream, it in fact
creates several early dialogs for which it should also handle transactions
such as PRACK (in case of reliable provisional responses), or UPDATEs and
such. 

 

So what do you think a B2B should do?

 

Thanks!

Ofer

 

 

_______________________________________________________________

If you lend someone $20, and never see that person again, it was probably
worth the investment. 

 

 


Classification: UNCLASSIFIED
Caveats: NONE



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From mary.ietf.barnes@gmail.com  Tue Feb 19 08:17:00 2013
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Subject: [sipcore] Fwd: Updates to draft-ietf-sipcore-rfc4244bis-11, version 1.
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Hi all,

This is a set of suggested changes from Dale based upon the comments
he sent to the list previously:
http://www.ietf.org/mail-archive/web/sipcore/current/msg05489.html

Responses are inline below [MB].   Based upon my responses, I do not
believe any of the changes impact normative behavior (as intended)
thus another WG and/or IETF LC should not be necessary, but I will
leave that decision to the chairs/AD.

Mary.


---------- Forwarded message ----------
From: Dale R. Worley <worley@ariadne.com>
Date: Wed, Feb 6, 2013 at 9:09 AM
Subject: Updates to draft-ietf-sipcore-rfc4244bis-11, version 1.
To: mary.h.barnes@gmail.com, pkyzivat@alum.mit.edu


Here are my proposed fixes to the -11 version.  I believe that all of
these changes are non-controverisial, in that they adjust the document
to express what we all agree it ought to mean.  But in a number of
places, the prescribed procedures are actually different from the
current text.  Please review this and get back to us with your
thoughts.

Thanks,

Dale

======================================================================

Updates to draft-ietf-sipcore-rfc4244bis-11, version 1.

This document is organized as a series of responses to the points in
my e-mail of 18 Jan
(http://www.ietf.org/mail-archive/web/sipcore/current/msg05489.html).

==========

> From: worley@ariadne.com (Dale R. Worley)
> To: SIPCORE <sipcore@ietf.org>
> Subject: [sipcore] Questions about draft-ietf-sipcore-rfc4244bis-11.txt
> Date: Fri, 18 Jan 2013 12:10:40 -0500 (EST)
>
> I've got a few questions about 4244bis.  I think that there are
> understood answers to most of them, but we might have to update the
> wording to remove ambiguity.

==========

> 1. When an entity receives a (non-100) provisional response to a
> request, should it modify the hi-entry by adding a Reason value
> carrying the status of the response?  The current text (section 10.2)
> clearly requires it.  If the request eventually receives a failure
> response, the 1xx Reason will (probably) later be replaced by the
> failure code.  (But not necessarily:  see section 10.2 paragraph 3.)
>
> The only example in draft-ietf-sipcore-rfc4244bis-callflows-01 of this
> situation is section 3.1 message F8, and the 180 status is not
> reported in a Reason header-value for <sip:office@example.com> (which
> thus contradicts section 10.2).
>
> The alternative is that 1xx responses do not set the Reason value
> (although they do add to the cache any new hi-entries they carry).
> This would require changing the text in section 9.3 step 2 and section
> 10.2.

NOTE:  I am assuming that non-100 provisional responses are intended
to carry hi-entries upstream, and that proxies that see those
responses are intended to cache the hi-entries carried in them.
(Those cached hi-entries can later be updated by a final response
which adds a Reason header value to them.)  If this is not so, these
changes need to be revised.

As far as I can tell, we have never intended for receipt of a
provisional (1xx) response to an INVITE to set a Reason header on the
corresponding cached hi-entry.  The only example in 4244bis-callflows
of this situation is message F6 of section 3.1, in which there would
be a Reason header on hi-entries 1.2 and 1.2.1 -- and there is not.
And RFC 4244, which also has the Reason header value mechanism, does
not specify that 1xx responses set Reason header values.

So I propose that we all agree that 1xx responses should not generate
Reason headers and we edit the text accordingly.
[MB] Correct.  That's been the intent for both 4244 and 4244bis.
It's mentioned in 9.4 that the HI is not sent in 100 responses (since
those do not result in retargeting), thus there would be no reason to
add a Reason header to anything.  [/MB]

 The changes are:

   section 9.3 step 2

   Step 2:  Add Reason header field

      The SIP entity adds one or more Reason header fields to the hi-
------^
change "The" to "If the response code is not 1xx, the"
      targeted-to-uri in the (newly) cached hi-entry reflecting the SIP
      response code in the non-100 response, per the procedures of
---------------^^^^^^^^^^^^^^^^^^^^^^^^^^^^
delete(?)
      Section 10.2.

This change results in:

      If the response code is not 1xx, the SIP entity adds one or more
      Reason header fields to the hi-targeted-to-uri in the (newly)
      cached hi-entry reflecting the SIP response, per the procedures
      of Section 10.2.

[MB] Note that the section is specific to non-100 responses, which is
why it is written as it currently is.
I don't see that your suggested text adds any more information - i.e.,
the IF isn't necessary.  We are stating specific behavior for that
specific step and an "IF" isn't necessary.  I do not think this change
is necessary. [/MB]

   10.2.  Reason in the History-Info Header Field

   A Reason header field is added to the "headers" component in an hi-
   targeted-to-uri when the hi-entry is added to the cache based upon
   the receipt of a non-100 or non-2xx SIP response, as described in
   Section 9.3.  If the Reason header field is being added due to
   receipt of an explicit SIP response and the response contains any
   Reason header fields (see [RFC3326]), then the SIP entity MUST
   include the Reason header fields in the "headers" component of the
   hi-targeted-to-uri in the last hi-entry added to the cache, unless
   the hi-targeted-to-uri is a Tel-URI.  If the SIP response does not
   contain a Reason header field, the SIP entity MUST include a Reason
   header field, containing the SIP Response Code, in the "headers"
   component of the hi-targeted-to-uri in the last hi-entry added to the
   cache, unless the hi-targeted-to-uri is a Tel-URI.

The changes to this paragraph are extensive enough that I am not
showing the individual differences.  Note I have changed "the last
hi-entry added to the cache" to "the corresponding hi-entry".  The
latter makes more sense to me, and specifies the same hi-entry.  But
others may prefer the current description, in which case we can retain
it.

The new version of this paragraph is:

   A Reason header field is added to the "headers" component in an hi-
   targeted-to-uri when a non-100, non-2xx SIP response is received to
   the corresponding INVITE, or when a response is received containing
   the hi-entry with a Reason header value, as described in Section
   9.3.
[MB] You have dropped the reference to the cache here and that very
important.  I think it's obvious that we
are dealing with the corresponding INVITE.  When we talk about SIP
responses, we do not qualify it as "the SIP response received to the
corresponding INVITE".  But, it certainly causes no harm.  I would
suggest that sentence read:
   A Reason header field is added to the "headers" component in an hi-
   targeted-to-uri when the hi-entry is added to the cache based upon
   the receipt of a non-100 or non-2xx SIP response to the
corresponding INVITE, as described in
   Section 9.3.

[/MB]

   If the Reason header value is being added to an hi-targeted-to-uri
   due to receipt of an explicit SIP response to the corresponding
   INVITE and the response contains any Reason header fields (see
   [RFC3326]), then the SIP entity MUST also include the Reason header
   fields in the "headers" component of the hi-targeted-to-uri in the
   corresponding cached hi-entry, unless the hi-targeted-to-uri is a
  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
   Tel-URI.  If the SIP response does not contain a Reason header
   field with a protocol of "SIP", the SIP entity MUST include a
   Reason header field, containing the SIP Response Code, in the
   "headers" component of the hi-targeted-to-uri in the corresponding
   cached hi-entry, unless the hi-targeted-to-uri is a Tel-URI.

[MB]  Note, you dropped "field" from the "Reason header value above,
which is not correct. So the real changes I see above are with regards
to highlighting that it's the corresponding cached hi-entry (rather
than the last.  I have annotated above that that is the only change
(AFAICT).  I can make that change as follows:
OLD:
    If the Reason header field is being added due to receipt of an explicit
    SIP response and the response contains any
    Reason header fields (see [RFC3326]), then the SIP entity MUST
    include the Reason header fields in the "headers" component of the
    hi-targeted-to-uri in the last hi-entry added to the cache, unless
    the hi-targeted-to-uri is a Tel-URI.

NEW:
   If the Reason header field is being added due to receipt of an explicit
   SIP response to the corresponding INVITE
                                    ^^^^^^^^^^^^^^^^^^^^^^
   and the response contains any
   Reason header fields (see [RFC3326]), then the SIP entity MUST
   include the Reason header fields in the "headers" component of the
   hi-targeted-to-uri in the corresponding cached hi-entry,  unless
                                         ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
   the hi-targeted-to-uri is a Tel-URI.
[/MB]

There is a slight change here in that if the response contains
"Reason: Q.850;cause=14", then both the Q.850 response code and the
SIP response code will be recorded in the cached hi-entry.  The
previous text caused only the Q.850 response code to be recorded.
[MB] I don't see that the previous text said that only the Q.850
response code was recorded.
The text says that "Reason header fields" as in plural fields are added. [/MB]

If a Reason header includes a reason-value with protocol "SIP", that
reason-value is recorded instead of the response code in the response.
(This is the same as the previous text.)

==========

> 2. During the life of the document, there was confusion about how
> hi-entries were to be organized.  The intention of the original writer
> and the current draft is that hi-entries are accumulated in a cache in
> the order the entity receives them (in requests and responses),
> allowing hi-entries that have the *same* index but *different* URIs,
> and (implicitly) eliminating duplicates.
[MB] It wasn't just the original author and the current author, this
has been the intent of
the document and the approach was agreed by the WG. Certainly, there were some
errors that might have lead one to believe otherwise, but that was
never the intent. [/MB]
>
> But some people who have worked on the draft (including myself) were
> for a while mistaken in believing that the hi-entries were to be
> organized into a tree, and hence the index values needed to be unique,
> and the History-Info header would list the hi-entries in tree-walking
> order (technically, "preorder").  The confusion was abetted by there
> being no examples that distinguish the two techniques.
[MB] I think the confusion is that if everyone supported HI, you would
get a tree.
You can still get a somewhat pruned tree using the indices.  I think
the confusion is that the RFC never
had any intention of defining application behavior.  It is merely
recording what happens at the SIP protocol level. That's it.  How an
application uses the information (whether to build a tree or whatever)
is out of scope. [/MB]
> Unfortunately, the -11 draft doesn't eliminate all the remnants of the
> mistake.  I haven't gone through -11 in detail, but "preorder" appears
> in section 9.2 and section 10.3, so at least those sections describe
> the accumulation of hi-entries incorrectly.

The fix to this issue overlaps the fix for issue #4:

> 4. If a downstream entity that does not support History-Info forks a
> request to entities that do support History-Info, an entity can
> receive two hi-entries with the same index that were not generated for
> the same fork of the request.  Conversely, an entity will receive
> responses that contain hi-entries that are duplicates of entries that
> are already in the cache.
>
> These facts imply that there is a process by which an entity
> determines whether a received hi-entry is "the same" as an hi-entry
> already in the cache (and so doesn't need to be added again).  Since
> index values cannot be depended upon to be unique, we really ought to
> state what the comparison rule is.
>
> I think it is sufficient to use the rule "the same index value and the
> URIs are equivalent (per RFC 3261 section 19.1.4, but ignoring the
> Reason header fields)".  This won't always give the result one would
> want (if a non-supporting downstream element forks the request to two
> target URIs that ultimately arrive at the same element, two hi-entries
> could be generated with the same index and URI), but this rule is
> simple, well-defined, and should work well in almost all situations.

In addition, when a response is received, it may contain hi-entries
that are duplicates of hi-entries already in the cache, except they
contain Reason header values that are not in the cached hi-entries.  I
believe we intend that the cached hi-entries be updated to include the
received Reason header values.

The edits to fix these three issues are:

   9.2.  Sending a Request with History-Info

   When sending a request, a SIP entity MUST include all the hi-entries
   from the cache that was created per Section 9.1.  In addition, the
   SIP entity MUST add a new hi-entry to the outgoing request, but the
   SIP entity MUST NOT add the hi-entry to the cache at this time.  The
--------------------------------------------------------------------^
Change this sentence to:  "The hi-entries in the outgoing request's
History-Info header are in the order of the hi-entries in the cache,
which is the order in which they were added to the cache."

[MB] That's fine. And, just for clarification, the statement below is
the OLD and the one above is the NEW [/MB]

   hi-entries in the outgoing request's History-Info header field is the
   preorder of the tree of hi-entries, that is, by the lexicographic
   ordering of the hi-indexes.  The new hi-entry is populated as
   follows:

   9.3.  Receiving a Response with History-Info or Request Timeouts

   Step 1:  Add hi-entry to cache

      The SIP entity MUST add the hi-entry that was added to the request
-----------------------------^
For clarity insert "to the cache" here, and...
      that received the non-100 response or timed out to the cache, if
------------------------------------------------------^^^^^^^^^^^^
... remove "to the cache" here.
      it was not already cached.  The hi-entry MUST be added to the
----------------------------------^
Change this sentence to "If so, the hi-entry MUST be added to the end
of the cache."

      cache in ascending order as indicated by the values in the hi-
      index parameters of the hi-entries (e.g., 1.2.1 comes after 1.2
      but before 1.2.2 or 1.3).

[MB] I can't decipher the changes you are suggesting above.  I think I
see your point.  However,
there is something very subtle here.  This is dealing with responses
and we add the hi-entries once we
get the response.  However, the value of the hi-index actually does
reflect a chronological order in terms of generating requests.  As
discussed previously, you lose that chronological ordering across
entities, but for a specific
SIP entity they do reflect the chronological processing at that
entity.  So, I don't think this is broken.  With the cache
model, we aren't adding them to the cache until we get the response
even though they were added to the request (in both chronological and
lexicographical order).
[/MB]


   9.3.  Receiving a Response with History-Info or Request Timeouts

   Step 3:  Add additional hi-entries

      The SIP entity MUST also add to the cache any hi-entries received
      in the response that are not already in the cache.  This situation
      can occur when the entity that generated the non-100 response
      retargeted the request before generating the response.  As per
      Step 1, the hi-entries MUST be added to the cache in ascending
      order as indicated by the values in the hi-index parameters of the
      hi-entries

This paragraph needs more extensive changes:
[MB] It's not clear to me why you think it needs to be done this way.
This is still mapping back to the original request sent by a specific
entity (e.g., Proxy A).   I think you are confused by the scenario
being described.  The retargeting was not done by Proxy A that is
caching the hi-entries.  It was done by Proxy B that received the
request and it retargeted before returning the 1xx response thus there
is an additional hi-entry that Proxy A should add to its cache. [/MB]

   Step 3:  Add and update additional hi-entries

      The SIP entity MUST also add to the cache any hi-entries received
      in the response that are not already in the cache.  This
      can occur when the entity that generated the non-100 response
      retargeted the request before generating the response.  Such
      hi-entries MUST be added to the end of the cache in the order
      they appear in the History-Info header field of the response.

      If an hi-entry in the response contains Reason header value(s)
      and the matching hi-entry in the cache does not, the Reason
      header value(s) MUST be added to the cached hi-entry.  This can
      occur when the hi-entry was added to the cache by a provisional
      response to an INVITE that was further retargeted, and later a
      final response to that INVITE is received.

      In these cases, an hi-entry in a response's History-Info header
      field and an hi-entry in the cache are considered matching when
      they have the same index value and the hi-targeted-to-uris are
      equivalent (per RFC 3261 section 19.1.4, but ignoring the Reason
      header fields).

   10.2.  Reason in the History-Info Header Field

   A Reason header field is added to the "headers" component in an hi-
   targeted-to-uri when the hi-entry is added to the cache based upon
----------------------------------------^^^^^^^^
change to "added to, or updated within,"

   the receipt of a non-100 or non-2xx SIP response, as described in
   Section 9.3.  [...]
[MB] Why "updated within"?  [/MB]

   10.3.  Indexing in the History-Info Header Field

   In order to maintain ordering and accurately reflect the retargeting
   of the request, the SIP entity MUST add a hi-index to each hi-entry.
   Per the syntax in Section 5, the hi-index consists of a series of
   nonnegative integer separated by dots (e.g., 1.1.2).  Each dot
   reflects a SIP forwarding hop.  The nonnegative integer following
   each dot reflects the order in which a request was retargeted at the
   hop.  The highest nonnegative integer at each hop reflects the number
   of entities to which the request has been retargeted at the specific
   hop (i.e., the number of branches) at the time that the request
   represented by this hi-entry was generated.  Thus, the indexing
   results in a logical tree representation for the history of the
   request and the hi-entries are given in the preorder of the tree.
-----------^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
delete this clause
[MB] Okay. [/MB]

==========

> 3. Due to the confusion discussed in #2, we have to consider whether
> consensus was actually found in the WGLC.
[MB] We had consensus in WGLC.  Your comments came well after WG last call
and after IETF LC.

>We thought we had
> consensus, but people did not have the same understanding of this
> aspect of the draft.  (Ugh.)

This is a procedural issue, not a technical one.  I don't see any
objection to correcting the normative text if everyone who cares
agrees that the changes express what we intended.  (We should check
this point with the AD.)

[MB] I do not see that we are changing any of the intended normative
behavior but rather fixing things that didn't quite reflect the
normative behavior as stated elsewhere (i.e., inconsistencies). [/MB]

==========

> 5. In draft-ietf-sipcore-rfc4244bis-callflows-01, examples 3.6 and 3.7
> do not show the History-Info value in the final 200 response to the
> UAC.  This final value would be useful, in particular, because they
> give further examples of how proxies process History-Info.

I was mistaken here:  The final 200 response from the example.com
proxy to the UAC in these examples has the same History-Info as the
200 response arriving at the example.com proxy, because there is no
History-Info information that the proxy knows that is not known to the
UAS that generated the 200 response.  Thus there is nothing
interesting to document.

==========

> 6. Interestingly, despite the fact that there can be duplicate index
> values, when a request is received by a UAS, there is no ambiguity in
> the history when tracing *back* from the last hi-entry (the one that
> denotes the UAS).  This is because even if this UAS is "down one fork"
> from a forking non-supporting proxy, none of the hi-entries generated
> "down the other fork" will be present in the request the UAS sees.
>
> This is in contrast to the situation at the UAC, which may see
> hi-entries with duplicate indexes, and the interpretation of "rc",
> "mp", and "np" parameters can be ambiguous.

This is an interesting observation, but it does not require
documentation in the RFCs.

==========

> 7. In example 3.6 message F6 in
> draft-ietf-sipcore-rfc4244bis-callflows-01, the History-Info is:
>
>    History-Info: <sip:bob@example.com>;index=1
>    History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302>;\
>                       index=1.1;rc=1
>    History-Info: <sip:carol@example.com>;index=1.2;mp=1
>    History-Info: <sip:carol@192.0.2.4?Reason=SIP%3Bcause%3D408>;\
>                       index=1.2.1;rc=1.2
>    History-Info: <sip:vm@example.com;\
>                       target=sip:bob%40example.com;cause=408>;\
>                       index=1.3;mp=1.2
>    History-Info: <sip:vm@192.0.2.6;\
>                       target=sip:bob%40example.com;cause=408>;\
>                       index=1.3.1;rc=1.3
>
> This is what we intend, as it shows both where the index=1.3 URI came
> from (sip:bob@example.com) as well as why the index=1.3 fork was
> generated (because sip:carol@example.com failed).  But it points out
> that we need to improve the description in rfc4244bis section 5 of
> what the values of the "rc" (etc.) parameters are.  A naive reader
> might think "rc=1.2" in the 1.3 entry means that
> <sip:vm@example.com;target=sip:bob%40example.com;cause=408> was
> generated as a contact for <sip:carol@192.0.2.4>.  Instead, it means
> that the 1.3 fork was generated *because* the 1.2 fork returned a
> response; it was generated *from* the index=1 URI.
>
> Actually, the real rules for interpreting "rc" (etc.) are not simple:
[MB] The above is an example, thus the description is not complete. [/MB]
>
> 1) If the index-val in the param points to an entry with a Reason
> header-param whose value is "SIP;cause=3xx" (after removing
> escaping!), then:
> - the URI was one of the Contact values in the 3xx response
> - that response was why the new fork was generated.
>
> 2) If the index-val in the param points to the parent of this
> hi-entry, then:
> - the URI was generated from the URI of the parent hi-entry
> - the fork was not generated due to the response from any other fork.
>
> 3) If the index-val in the param points to an hi-entry that is not the
> parent, and that hi-entry does not contain a Reason header-param
> containing a 3xx status, then:
> - the URI was generated from the URI of the parent hi-entry
> - the fork was generated due to the response from the hi-entry with
> the specified index-val.
>
> And this doesn't match the section 5 definition of the value of "rc"
> (etc.), which is:
>
>          The "rc" header field parameter contains the value of the
>          hi-index in the hi-entry with an hi-targeted-to-uri that
>          reflects the Request-URI that was retargeted

In section 5, these paragraphs need to be revised:

   o  hi-target-param: An optional parameter reflecting the mechanism by
      which the Request URI captured in the hi-targeted-to-uri in the
      History-Info header field value (hi-entry) was determined.  This
      parameter is either an "rc", "mp" or "np" header field parameter,
      which is interpreted as follows:

         "rc": The hi-targeted-to-URI represents a change in Request-URI
         while the target user remains the same.  This occurs for
         example when user has multiple AoRs as an alias.  The "rc"
         header field parameter contains the value of the hi-index in
         the hi-entry with an hi-targeted-to-uri that reflects the
         Request-URI that was retargeted

         "mp": The hi-targeted-to-URI represents a user other than the
         target user associated with the Request-URI in the incoming
         request that was retargeted.  This occurs when a request is
         statically or dynamically retargeted to another user
         represented by an AoR unassociated with the AoR of the original
         target user.  The "mp" header field parameter contains the
         value of the hi-index in the hi-entry with an hi-targeted-to-
         uri that reflects the Request-URI that was retargeted, thus
         identifying the "mapped from" target.

         "np": The hi-targeted-to-URI represents that there was no
         change in Request-URI.  This would apply for example when a
         proxy merely forwards a request to a next hop proxy and loose
         routing is used.  The "np" header field parameter contains the
         value of the hi-index in the hi-entry with an hi-targeted-to-
         uri that reflects the Request-URI that was copied unchanged
         into the request represented by this hi-entry.  That value will
         usually be the hi-index of the parent hi-entry of this hi-
         entry.

The revised paragraphs are:

[MB] I don't agree with this proposal - that's way too much detail for
this section.  I believe that the normative
sections describe how/when you add the "rc" tag which should match
what you are describing in your new bullets.    I will note that this
section has been updated several times both adding more text and
reducing the text in this section.  What you have proposed for the
bullets below (i.e., the "rc", "mp" etc.) is some of the level of
detail we had previously (in the -02).   I believe the text you
suggest is application logic - note that items 2 & 4 describe the "rc"
scenarios and this wasn't intended to be comprehensive.  If you can
highlight what's missing in those 2 bullets, we can perhaps make some
changes there.   I will note if these suggestions had been made much
earlier in the process, it would have been much more helpful - much of
the text you mention has been in the document since around the -02.
We did several rounds of major edits while developing this document
both while it was an individual and as a WG document.  Note that this
document became a WG document literally 3 years ago.  It's quite
disruptive to be raising this issues at this stage. [/MB]

   [This first paragraph is unchanged:]
   o  hi-target-param: An optional parameter reflecting the mechanism by
      which the Request URI captured in the hi-targeted-to-uri in the
      History-Info header field value (hi-entry) was determined.  This
      parameter is either an "rc", "mp" or "np" header field parameter,
      which is interpreted as follows:

         "rc": The hi-targeted-to-URI represents a change in Request-URI
         while the target user remains the same.  This occurs for
         example when user has multiple AoRs as an alias or when
         forwarding the request to a registered contact of an AoR.

         "mp": The hi-targeted-to-URI represents a user other than the
         target user associated with the Request-URI in the incoming
         request that was retargeted.  This occurs when a request is
         statically or dynamically retargeted to another user
         represented by an AoR unassociated with the AoR of the original
         target user.

         "np": The hi-targeted-to-URI represents that there was no
         change in Request-URI.  This would apply for example when a
         proxy merely forwards a request to a next hop proxy and loose
         routing is used.  The "np" header field parameter contains the
         value of the hi-index in the parent hi-entry of this hi-entry.

      If the index-val in the parameter points to the parent hi-entry of this
      hi-entry, then:
      - the URI was generated from the URI of the parent hi-entry
      - the fork was not generated due to the response from any other fork.

      If the index-val in the parameter points to an hi-entry with a Reason
      header value whose protocol is "SIP" and whose cause begins with "3",
      then:
      - the URI was one of the Contact values in the 3xx response
      - the fork was generated due to the 3xx response from the hi-entry with
      the specified index-val.

      If the index-val in the param points to an hi-entry that is not the
      parent, and that hi-entry does not contain a Reason header value
      containing a SIP 3xx cause, then:
      - the URI was generated from the URI of the parent hi-entry
      - the fork was generated due to the response from the hi-entry with
      the specified index-val.

======================================================================

I believe that is everything to be done.

Dale

From pkyzivat@alum.mit.edu  Tue Feb 19 08:23:42 2013
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Subject: Re: [sipcore] question about B2B and Forking
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On 2/19/13 8:02 AM, Ofer Goren wrote:
> Hi All.
>
> I know B2B behavior is not defined with too much details in the relevant
> SIP RFCs. I do ask for your opinion:
>
> In forking scenarios, if B2B forks an INVITE to several UAS-es, should
> it forward downstream all the provisional responses to the generating
> UAC? Or, should it forward only one,” absorbing” the rest? Should it
> even wait for provisional responses, or should it generates its own
> toward the UAC, independent to the upstream side of the session?
>
> If it should forward all provisional responses downstream, it in fact
> creates several early dialogs for which it should also handle
> transactions such as PRACK (in case of reliable provisional responses),
> or UPDATEs and such.
>
> So what do you think a B2B should do?

These are all design decisions for you when building your B2BUA.
The answers depend upon your goals/objectives.

	Good Luck,
	Paul


From pkyzivat@alum.mit.edu  Tue Feb 19 08:50:20 2013
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Subject: Re: [sipcore] Fwd: Updates to draft-ietf-sipcore-rfc4244bis-11, version 1.
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Mary & Dale,

This is very hard to follow.
But I do get it that the two of you are getting closer but still not in 
total agreement. Can the two of you reach a mutually satisfactory 
conclusion?

I'd love to have a new version with a direct diff to evaluate it all.

I must say that I am surprised that so much effort was put into the 
hierarchical numbering scheme for entries and yet the conclusion is that 
the entries are not sorted in accord with this hierarchy. (Or did I get 
that wrong?)

	Thanks,
	Paul

On 2/19/13 11:16 AM, Mary Barnes wrote:
> Hi all,
>
> This is a set of suggested changes from Dale based upon the comments
> he sent to the list previously:
> http://www.ietf.org/mail-archive/web/sipcore/current/msg05489.html
>
> Responses are inline below [MB].   Based upon my responses, I do not
> believe any of the changes impact normative behavior (as intended)
> thus another WG and/or IETF LC should not be necessary, but I will
> leave that decision to the chairs/AD.
>
> Mary.
>
>
> ---------- Forwarded message ----------
> From: Dale R. Worley <worley@ariadne.com>
> Date: Wed, Feb 6, 2013 at 9:09 AM
> Subject: Updates to draft-ietf-sipcore-rfc4244bis-11, version 1.
> To: mary.h.barnes@gmail.com, pkyzivat@alum.mit.edu
>
>
> Here are my proposed fixes to the -11 version.  I believe that all of
> these changes are non-controverisial, in that they adjust the document
> to express what we all agree it ought to mean.  But in a number of
> places, the prescribed procedures are actually different from the
> current text.  Please review this and get back to us with your
> thoughts.
>
> Thanks,
>
> Dale
>
> ======================================================================
>
> Updates to draft-ietf-sipcore-rfc4244bis-11, version 1.
>
> This document is organized as a series of responses to the points in
> my e-mail of 18 Jan
> (http://www.ietf.org/mail-archive/web/sipcore/current/msg05489.html).
>
> ==========
>
>> From: worley@ariadne.com (Dale R. Worley)
>> To: SIPCORE <sipcore@ietf.org>
>> Subject: [sipcore] Questions about draft-ietf-sipcore-rfc4244bis-11.txt
>> Date: Fri, 18 Jan 2013 12:10:40 -0500 (EST)
>>
>> I've got a few questions about 4244bis.  I think that there are
>> understood answers to most of them, but we might have to update the
>> wording to remove ambiguity.
>
> ==========
>
>> 1. When an entity receives a (non-100) provisional response to a
>> request, should it modify the hi-entry by adding a Reason value
>> carrying the status of the response?  The current text (section 10.2)
>> clearly requires it.  If the request eventually receives a failure
>> response, the 1xx Reason will (probably) later be replaced by the
>> failure code.  (But not necessarily:  see section 10.2 paragraph 3.)
>>
>> The only example in draft-ietf-sipcore-rfc4244bis-callflows-01 of this
>> situation is section 3.1 message F8, and the 180 status is not
>> reported in a Reason header-value for <sip:office@example.com> (which
>> thus contradicts section 10.2).
>>
>> The alternative is that 1xx responses do not set the Reason value
>> (although they do add to the cache any new hi-entries they carry).
>> This would require changing the text in section 9.3 step 2 and section
>> 10.2.
>
> NOTE:  I am assuming that non-100 provisional responses are intended
> to carry hi-entries upstream, and that proxies that see those
> responses are intended to cache the hi-entries carried in them.
> (Those cached hi-entries can later be updated by a final response
> which adds a Reason header value to them.)  If this is not so, these
> changes need to be revised.
>
> As far as I can tell, we have never intended for receipt of a
> provisional (1xx) response to an INVITE to set a Reason header on the
> corresponding cached hi-entry.  The only example in 4244bis-callflows
> of this situation is message F6 of section 3.1, in which there would
> be a Reason header on hi-entries 1.2 and 1.2.1 -- and there is not.
> And RFC 4244, which also has the Reason header value mechanism, does
> not specify that 1xx responses set Reason header values.
>
> So I propose that we all agree that 1xx responses should not generate
> Reason headers and we edit the text accordingly.
> [MB] Correct.  That's been the intent for both 4244 and 4244bis.
> It's mentioned in 9.4 that the HI is not sent in 100 responses (since
> those do not result in retargeting), thus there would be no reason to
> add a Reason header to anything.  [/MB]
>
>   The changes are:
>
>     section 9.3 step 2
>
>     Step 2:  Add Reason header field
>
>        The SIP entity adds one or more Reason header fields to the hi-
> ------^
> change "The" to "If the response code is not 1xx, the"
>        targeted-to-uri in the (newly) cached hi-entry reflecting the SIP
>        response code in the non-100 response, per the procedures of
> ---------------^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> delete(?)
>        Section 10.2.
>
> This change results in:
>
>        If the response code is not 1xx, the SIP entity adds one or more
>        Reason header fields to the hi-targeted-to-uri in the (newly)
>        cached hi-entry reflecting the SIP response, per the procedures
>        of Section 10.2.
>
> [MB] Note that the section is specific to non-100 responses, which is
> why it is written as it currently is.
> I don't see that your suggested text adds any more information - i.e.,
> the IF isn't necessary.  We are stating specific behavior for that
> specific step and an "IF" isn't necessary.  I do not think this change
> is necessary. [/MB]
>
>     10.2.  Reason in the History-Info Header Field
>
>     A Reason header field is added to the "headers" component in an hi-
>     targeted-to-uri when the hi-entry is added to the cache based upon
>     the receipt of a non-100 or non-2xx SIP response, as described in
>     Section 9.3.  If the Reason header field is being added due to
>     receipt of an explicit SIP response and the response contains any
>     Reason header fields (see [RFC3326]), then the SIP entity MUST
>     include the Reason header fields in the "headers" component of the
>     hi-targeted-to-uri in the last hi-entry added to the cache, unless
>     the hi-targeted-to-uri is a Tel-URI.  If the SIP response does not
>     contain a Reason header field, the SIP entity MUST include a Reason
>     header field, containing the SIP Response Code, in the "headers"
>     component of the hi-targeted-to-uri in the last hi-entry added to the
>     cache, unless the hi-targeted-to-uri is a Tel-URI.
>
> The changes to this paragraph are extensive enough that I am not
> showing the individual differences.  Note I have changed "the last
> hi-entry added to the cache" to "the corresponding hi-entry".  The
> latter makes more sense to me, and specifies the same hi-entry.  But
> others may prefer the current description, in which case we can retain
> it.
>
> The new version of this paragraph is:
>
>     A Reason header field is added to the "headers" component in an hi-
>     targeted-to-uri when a non-100, non-2xx SIP response is received to
>     the corresponding INVITE, or when a response is received containing
>     the hi-entry with a Reason header value, as described in Section
>     9.3.
> [MB] You have dropped the reference to the cache here and that very
> important.  I think it's obvious that we
> are dealing with the corresponding INVITE.  When we talk about SIP
> responses, we do not qualify it as "the SIP response received to the
> corresponding INVITE".  But, it certainly causes no harm.  I would
> suggest that sentence read:
>     A Reason header field is added to the "headers" component in an hi-
>     targeted-to-uri when the hi-entry is added to the cache based upon
>     the receipt of a non-100 or non-2xx SIP response to the
> corresponding INVITE, as described in
>     Section 9.3.
>
> [/MB]
>
>     If the Reason header value is being added to an hi-targeted-to-uri
>     due to receipt of an explicit SIP response to the corresponding
>     INVITE and the response contains any Reason header fields (see
>     [RFC3326]), then the SIP entity MUST also include the Reason header
>     fields in the "headers" component of the hi-targeted-to-uri in the
>     corresponding cached hi-entry, unless the hi-targeted-to-uri is a
>    ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
>     Tel-URI.  If the SIP response does not contain a Reason header
>     field with a protocol of "SIP", the SIP entity MUST include a
>     Reason header field, containing the SIP Response Code, in the
>     "headers" component of the hi-targeted-to-uri in the corresponding
>     cached hi-entry, unless the hi-targeted-to-uri is a Tel-URI.
>
> [MB]  Note, you dropped "field" from the "Reason header value above,
> which is not correct. So the real changes I see above are with regards
> to highlighting that it's the corresponding cached hi-entry (rather
> than the last.  I have annotated above that that is the only change
> (AFAICT).  I can make that change as follows:
> OLD:
>      If the Reason header field is being added due to receipt of an explicit
>      SIP response and the response contains any
>      Reason header fields (see [RFC3326]), then the SIP entity MUST
>      include the Reason header fields in the "headers" component of the
>      hi-targeted-to-uri in the last hi-entry added to the cache, unless
>      the hi-targeted-to-uri is a Tel-URI.
>
> NEW:
>     If the Reason header field is being added due to receipt of an explicit
>     SIP response to the corresponding INVITE
>                                      ^^^^^^^^^^^^^^^^^^^^^^
>     and the response contains any
>     Reason header fields (see [RFC3326]), then the SIP entity MUST
>     include the Reason header fields in the "headers" component of the
>     hi-targeted-to-uri in the corresponding cached hi-entry,  unless
>                                           ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
>     the hi-targeted-to-uri is a Tel-URI.
> [/MB]
>
> There is a slight change here in that if the response contains
> "Reason: Q.850;cause=14", then both the Q.850 response code and the
> SIP response code will be recorded in the cached hi-entry.  The
> previous text caused only the Q.850 response code to be recorded.
> [MB] I don't see that the previous text said that only the Q.850
> response code was recorded.
> The text says that "Reason header fields" as in plural fields are added. [/MB]
>
> If a Reason header includes a reason-value with protocol "SIP", that
> reason-value is recorded instead of the response code in the response.
> (This is the same as the previous text.)
>
> ==========
>
>> 2. During the life of the document, there was confusion about how
>> hi-entries were to be organized.  The intention of the original writer
>> and the current draft is that hi-entries are accumulated in a cache in
>> the order the entity receives them (in requests and responses),
>> allowing hi-entries that have the *same* index but *different* URIs,
>> and (implicitly) eliminating duplicates.
> [MB] It wasn't just the original author and the current author, this
> has been the intent of
> the document and the approach was agreed by the WG. Certainly, there were some
> errors that might have lead one to believe otherwise, but that was
> never the intent. [/MB]
>>
>> But some people who have worked on the draft (including myself) were
>> for a while mistaken in believing that the hi-entries were to be
>> organized into a tree, and hence the index values needed to be unique,
>> and the History-Info header would list the hi-entries in tree-walking
>> order (technically, "preorder").  The confusion was abetted by there
>> being no examples that distinguish the two techniques.
> [MB] I think the confusion is that if everyone supported HI, you would
> get a tree.
> You can still get a somewhat pruned tree using the indices.  I think
> the confusion is that the RFC never
> had any intention of defining application behavior.  It is merely
> recording what happens at the SIP protocol level. That's it.  How an
> application uses the information (whether to build a tree or whatever)
> is out of scope. [/MB]
>> Unfortunately, the -11 draft doesn't eliminate all the remnants of the
>> mistake.  I haven't gone through -11 in detail, but "preorder" appears
>> in section 9.2 and section 10.3, so at least those sections describe
>> the accumulation of hi-entries incorrectly.
>
> The fix to this issue overlaps the fix for issue #4:
>
>> 4. If a downstream entity that does not support History-Info forks a
>> request to entities that do support History-Info, an entity can
>> receive two hi-entries with the same index that were not generated for
>> the same fork of the request.  Conversely, an entity will receive
>> responses that contain hi-entries that are duplicates of entries that
>> are already in the cache.
>>
>> These facts imply that there is a process by which an entity
>> determines whether a received hi-entry is "the same" as an hi-entry
>> already in the cache (and so doesn't need to be added again).  Since
>> index values cannot be depended upon to be unique, we really ought to
>> state what the comparison rule is.
>>
>> I think it is sufficient to use the rule "the same index value and the
>> URIs are equivalent (per RFC 3261 section 19.1.4, but ignoring the
>> Reason header fields)".  This won't always give the result one would
>> want (if a non-supporting downstream element forks the request to two
>> target URIs that ultimately arrive at the same element, two hi-entries
>> could be generated with the same index and URI), but this rule is
>> simple, well-defined, and should work well in almost all situations.
>
> In addition, when a response is received, it may contain hi-entries
> that are duplicates of hi-entries already in the cache, except they
> contain Reason header values that are not in the cached hi-entries.  I
> believe we intend that the cached hi-entries be updated to include the
> received Reason header values.
>
> The edits to fix these three issues are:
>
>     9.2.  Sending a Request with History-Info
>
>     When sending a request, a SIP entity MUST include all the hi-entries
>     from the cache that was created per Section 9.1.  In addition, the
>     SIP entity MUST add a new hi-entry to the outgoing request, but the
>     SIP entity MUST NOT add the hi-entry to the cache at this time.  The
> --------------------------------------------------------------------^
> Change this sentence to:  "The hi-entries in the outgoing request's
> History-Info header are in the order of the hi-entries in the cache,
> which is the order in which they were added to the cache."
>
> [MB] That's fine. And, just for clarification, the statement below is
> the OLD and the one above is the NEW [/MB]
>
>     hi-entries in the outgoing request's History-Info header field is the
>     preorder of the tree of hi-entries, that is, by the lexicographic
>     ordering of the hi-indexes.  The new hi-entry is populated as
>     follows:
>
>     9.3.  Receiving a Response with History-Info or Request Timeouts
>
>     Step 1:  Add hi-entry to cache
>
>        The SIP entity MUST add the hi-entry that was added to the request
> -----------------------------^
> For clarity insert "to the cache" here, and...
>        that received the non-100 response or timed out to the cache, if
> ------------------------------------------------------^^^^^^^^^^^^
> ... remove "to the cache" here.
>        it was not already cached.  The hi-entry MUST be added to the
> ----------------------------------^
> Change this sentence to "If so, the hi-entry MUST be added to the end
> of the cache."
>
>        cache in ascending order as indicated by the values in the hi-
>        index parameters of the hi-entries (e.g., 1.2.1 comes after 1.2
>        but before 1.2.2 or 1.3).
>
> [MB] I can't decipher the changes you are suggesting above.  I think I
> see your point.  However,
> there is something very subtle here.  This is dealing with responses
> and we add the hi-entries once we
> get the response.  However, the value of the hi-index actually does
> reflect a chronological order in terms of generating requests.  As
> discussed previously, you lose that chronological ordering across
> entities, but for a specific
> SIP entity they do reflect the chronological processing at that
> entity.  So, I don't think this is broken.  With the cache
> model, we aren't adding them to the cache until we get the response
> even though they were added to the request (in both chronological and
> lexicographical order).
> [/MB]
>
>
>     9.3.  Receiving a Response with History-Info or Request Timeouts
>
>     Step 3:  Add additional hi-entries
>
>        The SIP entity MUST also add to the cache any hi-entries received
>        in the response that are not already in the cache.  This situation
>        can occur when the entity that generated the non-100 response
>        retargeted the request before generating the response.  As per
>        Step 1, the hi-entries MUST be added to the cache in ascending
>        order as indicated by the values in the hi-index parameters of the
>        hi-entries
>
> This paragraph needs more extensive changes:
> [MB] It's not clear to me why you think it needs to be done this way.
> This is still mapping back to the original request sent by a specific
> entity (e.g., Proxy A).   I think you are confused by the scenario
> being described.  The retargeting was not done by Proxy A that is
> caching the hi-entries.  It was done by Proxy B that received the
> request and it retargeted before returning the 1xx response thus there
> is an additional hi-entry that Proxy A should add to its cache. [/MB]
>
>     Step 3:  Add and update additional hi-entries
>
>        The SIP entity MUST also add to the cache any hi-entries received
>        in the response that are not already in the cache.  This
>        can occur when the entity that generated the non-100 response
>        retargeted the request before generating the response.  Such
>        hi-entries MUST be added to the end of the cache in the order
>        they appear in the History-Info header field of the response.
>
>        If an hi-entry in the response contains Reason header value(s)
>        and the matching hi-entry in the cache does not, the Reason
>        header value(s) MUST be added to the cached hi-entry.  This can
>        occur when the hi-entry was added to the cache by a provisional
>        response to an INVITE that was further retargeted, and later a
>        final response to that INVITE is received.
>
>        In these cases, an hi-entry in a response's History-Info header
>        field and an hi-entry in the cache are considered matching when
>        they have the same index value and the hi-targeted-to-uris are
>        equivalent (per RFC 3261 section 19.1.4, but ignoring the Reason
>        header fields).
>
>     10.2.  Reason in the History-Info Header Field
>
>     A Reason header field is added to the "headers" component in an hi-
>     targeted-to-uri when the hi-entry is added to the cache based upon
> ----------------------------------------^^^^^^^^
> change to "added to, or updated within,"
>
>     the receipt of a non-100 or non-2xx SIP response, as described in
>     Section 9.3.  [...]
> [MB] Why "updated within"?  [/MB]
>
>     10.3.  Indexing in the History-Info Header Field
>
>     In order to maintain ordering and accurately reflect the retargeting
>     of the request, the SIP entity MUST add a hi-index to each hi-entry.
>     Per the syntax in Section 5, the hi-index consists of a series of
>     nonnegative integer separated by dots (e.g., 1.1.2).  Each dot
>     reflects a SIP forwarding hop.  The nonnegative integer following
>     each dot reflects the order in which a request was retargeted at the
>     hop.  The highest nonnegative integer at each hop reflects the number
>     of entities to which the request has been retargeted at the specific
>     hop (i.e., the number of branches) at the time that the request
>     represented by this hi-entry was generated.  Thus, the indexing
>     results in a logical tree representation for the history of the
>     request and the hi-entries are given in the preorder of the tree.
> -----------^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> delete this clause
> [MB] Okay. [/MB]
>
> ==========
>
>> 3. Due to the confusion discussed in #2, we have to consider whether
>> consensus was actually found in the WGLC.
> [MB] We had consensus in WGLC.  Your comments came well after WG last call
> and after IETF LC.
>
>> We thought we had
>> consensus, but people did not have the same understanding of this
>> aspect of the draft.  (Ugh.)
>
> This is a procedural issue, not a technical one.  I don't see any
> objection to correcting the normative text if everyone who cares
> agrees that the changes express what we intended.  (We should check
> this point with the AD.)
>
> [MB] I do not see that we are changing any of the intended normative
> behavior but rather fixing things that didn't quite reflect the
> normative behavior as stated elsewhere (i.e., inconsistencies). [/MB]
>
> ==========
>
>> 5. In draft-ietf-sipcore-rfc4244bis-callflows-01, examples 3.6 and 3.7
>> do not show the History-Info value in the final 200 response to the
>> UAC.  This final value would be useful, in particular, because they
>> give further examples of how proxies process History-Info.
>
> I was mistaken here:  The final 200 response from the example.com
> proxy to the UAC in these examples has the same History-Info as the
> 200 response arriving at the example.com proxy, because there is no
> History-Info information that the proxy knows that is not known to the
> UAS that generated the 200 response.  Thus there is nothing
> interesting to document.
>
> ==========
>
>> 6. Interestingly, despite the fact that there can be duplicate index
>> values, when a request is received by a UAS, there is no ambiguity in
>> the history when tracing *back* from the last hi-entry (the one that
>> denotes the UAS).  This is because even if this UAS is "down one fork"
>> from a forking non-supporting proxy, none of the hi-entries generated
>> "down the other fork" will be present in the request the UAS sees.
>>
>> This is in contrast to the situation at the UAC, which may see
>> hi-entries with duplicate indexes, and the interpretation of "rc",
>> "mp", and "np" parameters can be ambiguous.
>
> This is an interesting observation, but it does not require
> documentation in the RFCs.
>
> ==========
>
>> 7. In example 3.6 message F6 in
>> draft-ietf-sipcore-rfc4244bis-callflows-01, the History-Info is:
>>
>>     History-Info: <sip:bob@example.com>;index=1
>>     History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302>;\
>>                        index=1.1;rc=1
>>     History-Info: <sip:carol@example.com>;index=1.2;mp=1
>>     History-Info: <sip:carol@192.0.2.4?Reason=SIP%3Bcause%3D408>;\
>>                        index=1.2.1;rc=1.2
>>     History-Info: <sip:vm@example.com;\
>>                        target=sip:bob%40example.com;cause=408>;\
>>                        index=1.3;mp=1.2
>>     History-Info: <sip:vm@192.0.2.6;\
>>                        target=sip:bob%40example.com;cause=408>;\
>>                        index=1.3.1;rc=1.3
>>
>> This is what we intend, as it shows both where the index=1.3 URI came
>> from (sip:bob@example.com) as well as why the index=1.3 fork was
>> generated (because sip:carol@example.com failed).  But it points out
>> that we need to improve the description in rfc4244bis section 5 of
>> what the values of the "rc" (etc.) parameters are.  A naive reader
>> might think "rc=1.2" in the 1.3 entry means that
>> <sip:vm@example.com;target=sip:bob%40example.com;cause=408> was
>> generated as a contact for <sip:carol@192.0.2.4>.  Instead, it means
>> that the 1.3 fork was generated *because* the 1.2 fork returned a
>> response; it was generated *from* the index=1 URI.
>>
>> Actually, the real rules for interpreting "rc" (etc.) are not simple:
> [MB] The above is an example, thus the description is not complete. [/MB]
>>
>> 1) If the index-val in the param points to an entry with a Reason
>> header-param whose value is "SIP;cause=3xx" (after removing
>> escaping!), then:
>> - the URI was one of the Contact values in the 3xx response
>> - that response was why the new fork was generated.
>>
>> 2) If the index-val in the param points to the parent of this
>> hi-entry, then:
>> - the URI was generated from the URI of the parent hi-entry
>> - the fork was not generated due to the response from any other fork.
>>
>> 3) If the index-val in the param points to an hi-entry that is not the
>> parent, and that hi-entry does not contain a Reason header-param
>> containing a 3xx status, then:
>> - the URI was generated from the URI of the parent hi-entry
>> - the fork was generated due to the response from the hi-entry with
>> the specified index-val.
>>
>> And this doesn't match the section 5 definition of the value of "rc"
>> (etc.), which is:
>>
>>           The "rc" header field parameter contains the value of the
>>           hi-index in the hi-entry with an hi-targeted-to-uri that
>>           reflects the Request-URI that was retargeted
>
> In section 5, these paragraphs need to be revised:
>
>     o  hi-target-param: An optional parameter reflecting the mechanism by
>        which the Request URI captured in the hi-targeted-to-uri in the
>        History-Info header field value (hi-entry) was determined.  This
>        parameter is either an "rc", "mp" or "np" header field parameter,
>        which is interpreted as follows:
>
>           "rc": The hi-targeted-to-URI represents a change in Request-URI
>           while the target user remains the same.  This occurs for
>           example when user has multiple AoRs as an alias.  The "rc"
>           header field parameter contains the value of the hi-index in
>           the hi-entry with an hi-targeted-to-uri that reflects the
>           Request-URI that was retargeted
>
>           "mp": The hi-targeted-to-URI represents a user other than the
>           target user associated with the Request-URI in the incoming
>           request that was retargeted.  This occurs when a request is
>           statically or dynamically retargeted to another user
>           represented by an AoR unassociated with the AoR of the original
>           target user.  The "mp" header field parameter contains the
>           value of the hi-index in the hi-entry with an hi-targeted-to-
>           uri that reflects the Request-URI that was retargeted, thus
>           identifying the "mapped from" target.
>
>           "np": The hi-targeted-to-URI represents that there was no
>           change in Request-URI.  This would apply for example when a
>           proxy merely forwards a request to a next hop proxy and loose
>           routing is used.  The "np" header field parameter contains the
>           value of the hi-index in the hi-entry with an hi-targeted-to-
>           uri that reflects the Request-URI that was copied unchanged
>           into the request represented by this hi-entry.  That value will
>           usually be the hi-index of the parent hi-entry of this hi-
>           entry.
>
> The revised paragraphs are:
>
> [MB] I don't agree with this proposal - that's way too much detail for
> this section.  I believe that the normative
> sections describe how/when you add the "rc" tag which should match
> what you are describing in your new bullets.    I will note that this
> section has been updated several times both adding more text and
> reducing the text in this section.  What you have proposed for the
> bullets below (i.e., the "rc", "mp" etc.) is some of the level of
> detail we had previously (in the -02).   I believe the text you
> suggest is application logic - note that items 2 & 4 describe the "rc"
> scenarios and this wasn't intended to be comprehensive.  If you can
> highlight what's missing in those 2 bullets, we can perhaps make some
> changes there.   I will note if these suggestions had been made much
> earlier in the process, it would have been much more helpful - much of
> the text you mention has been in the document since around the -02.
> We did several rounds of major edits while developing this document
> both while it was an individual and as a WG document.  Note that this
> document became a WG document literally 3 years ago.  It's quite
> disruptive to be raising this issues at this stage. [/MB]
>
>     [This first paragraph is unchanged:]
>     o  hi-target-param: An optional parameter reflecting the mechanism by
>        which the Request URI captured in the hi-targeted-to-uri in the
>        History-Info header field value (hi-entry) was determined.  This
>        parameter is either an "rc", "mp" or "np" header field parameter,
>        which is interpreted as follows:
>
>           "rc": The hi-targeted-to-URI represents a change in Request-URI
>           while the target user remains the same.  This occurs for
>           example when user has multiple AoRs as an alias or when
>           forwarding the request to a registered contact of an AoR.
>
>           "mp": The hi-targeted-to-URI represents a user other than the
>           target user associated with the Request-URI in the incoming
>           request that was retargeted.  This occurs when a request is
>           statically or dynamically retargeted to another user
>           represented by an AoR unassociated with the AoR of the original
>           target user.
>
>           "np": The hi-targeted-to-URI represents that there was no
>           change in Request-URI.  This would apply for example when a
>           proxy merely forwards a request to a next hop proxy and loose
>           routing is used.  The "np" header field parameter contains the
>           value of the hi-index in the parent hi-entry of this hi-entry.
>
>        If the index-val in the parameter points to the parent hi-entry of this
>        hi-entry, then:
>        - the URI was generated from the URI of the parent hi-entry
>        - the fork was not generated due to the response from any other fork.
>
>        If the index-val in the parameter points to an hi-entry with a Reason
>        header value whose protocol is "SIP" and whose cause begins with "3",
>        then:
>        - the URI was one of the Contact values in the 3xx response
>        - the fork was generated due to the 3xx response from the hi-entry with
>        the specified index-val.
>
>        If the index-val in the param points to an hi-entry that is not the
>        parent, and that hi-entry does not contain a Reason header value
>        containing a SIP 3xx cause, then:
>        - the URI was generated from the URI of the parent hi-entry
>        - the fork was generated due to the response from the hi-entry with
>        the specified index-val.
>
> ======================================================================
>
> I believe that is everything to be done.
>
> Dale
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>


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References: <201302061509.r16F90sh016532@freeze.ariadne.com> <CABmDk8ksvP7rL-NRu1wFiRx+ZB9ki5ENrRzmOoyS=DUjWvzpGw@mail.gmail.com> <CAHBDyN7j0aXO6G03TJiuCPjRTrNyHhgBJAAnauk+TYfS06CCBA@mail.gmail.com> <5123AD43.2080400@alum.mit.edu>
Date: Tue, 19 Feb 2013 10:55:48 -0600
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From: Mary Barnes <mary.ietf.barnes@gmail.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Fwd: Updates to draft-ietf-sipcore-rfc4244bis-11, version 1.
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On Tue, Feb 19, 2013 at 10:50 AM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
> Mary & Dale,
>
> This is very hard to follow.
> But I do get it that the two of you are getting closer but still not in
> total agreement. Can the two of you reach a mutually satisfactory
> conclusion?
>
> I'd love to have a new version with a direct diff to evaluate it all.
>
> I must say that I am surprised that so much effort was put into the
> hierarchical numbering scheme for entries and yet the conclusion is that the
> entries are not sorted in accord with this hierarchy. (Or did I get that
> wrong?)

[MB] As I highlighted to Dale, within an entity they are in order.
The issue is due to the fact that not all entities will support HI,
thus you cannot guarantee that the lexicographical order is meaningful
(amongst all the entities).  Since we do require an entity to add the
entries in chronological order using the indexing scheme as specifed
they are in order for that entity.  When the request is forwarded and
a response contains additional hi-entires (due to retargeting at the
next hop), the ordering is guaranteed for the entity to which the
request was forwarded.  The issue comes when you get a reset to 0
indicating an entity did not support HI, then you lose any ordering.
[/MB]
>
>         Thanks,
>         Paul
>
>
> On 2/19/13 11:16 AM, Mary Barnes wrote:
>>
>> Hi all,
>>
>> This is a set of suggested changes from Dale based upon the comments
>> he sent to the list previously:
>> http://www.ietf.org/mail-archive/web/sipcore/current/msg05489.html
>>
>> Responses are inline below [MB].   Based upon my responses, I do not
>> believe any of the changes impact normative behavior (as intended)
>> thus another WG and/or IETF LC should not be necessary, but I will
>> leave that decision to the chairs/AD.
>>
>> Mary.
>>
>>
>> ---------- Forwarded message ----------
>> From: Dale R. Worley <worley@ariadne.com>
>> Date: Wed, Feb 6, 2013 at 9:09 AM
>> Subject: Updates to draft-ietf-sipcore-rfc4244bis-11, version 1.
>> To: mary.h.barnes@gmail.com, pkyzivat@alum.mit.edu
>>
>>
>> Here are my proposed fixes to the -11 version.  I believe that all of
>> these changes are non-controverisial, in that they adjust the document
>> to express what we all agree it ought to mean.  But in a number of
>> places, the prescribed procedures are actually different from the
>> current text.  Please review this and get back to us with your
>> thoughts.
>>
>> Thanks,
>>
>> Dale
>>
>> ======================================================================
>>
>> Updates to draft-ietf-sipcore-rfc4244bis-11, version 1.
>>
>> This document is organized as a series of responses to the points in
>> my e-mail of 18 Jan
>> (http://www.ietf.org/mail-archive/web/sipcore/current/msg05489.html).
>>
>> ==========
>>
>>> From: worley@ariadne.com (Dale R. Worley)
>>> To: SIPCORE <sipcore@ietf.org>
>>> Subject: [sipcore] Questions about draft-ietf-sipcore-rfc4244bis-11.txt
>>> Date: Fri, 18 Jan 2013 12:10:40 -0500 (EST)
>>>
>>> I've got a few questions about 4244bis.  I think that there are
>>> understood answers to most of them, but we might have to update the
>>> wording to remove ambiguity.
>>
>>
>> ==========
>>
>>> 1. When an entity receives a (non-100) provisional response to a
>>> request, should it modify the hi-entry by adding a Reason value
>>> carrying the status of the response?  The current text (section 10.2)
>>> clearly requires it.  If the request eventually receives a failure
>>> response, the 1xx Reason will (probably) later be replaced by the
>>> failure code.  (But not necessarily:  see section 10.2 paragraph 3.)
>>>
>>> The only example in draft-ietf-sipcore-rfc4244bis-callflows-01 of this
>>> situation is section 3.1 message F8, and the 180 status is not
>>> reported in a Reason header-value for <sip:office@example.com> (which
>>> thus contradicts section 10.2).
>>>
>>> The alternative is that 1xx responses do not set the Reason value
>>> (although they do add to the cache any new hi-entries they carry).
>>> This would require changing the text in section 9.3 step 2 and section
>>> 10.2.
>>
>>
>> NOTE:  I am assuming that non-100 provisional responses are intended
>> to carry hi-entries upstream, and that proxies that see those
>> responses are intended to cache the hi-entries carried in them.
>> (Those cached hi-entries can later be updated by a final response
>> which adds a Reason header value to them.)  If this is not so, these
>> changes need to be revised.
>>
>> As far as I can tell, we have never intended for receipt of a
>> provisional (1xx) response to an INVITE to set a Reason header on the
>> corresponding cached hi-entry.  The only example in 4244bis-callflows
>> of this situation is message F6 of section 3.1, in which there would
>> be a Reason header on hi-entries 1.2 and 1.2.1 -- and there is not.
>> And RFC 4244, which also has the Reason header value mechanism, does
>> not specify that 1xx responses set Reason header values.
>>
>> So I propose that we all agree that 1xx responses should not generate
>> Reason headers and we edit the text accordingly.
>> [MB] Correct.  That's been the intent for both 4244 and 4244bis.
>> It's mentioned in 9.4 that the HI is not sent in 100 responses (since
>> those do not result in retargeting), thus there would be no reason to
>> add a Reason header to anything.  [/MB]
>>
>>   The changes are:
>>
>>     section 9.3 step 2
>>
>>     Step 2:  Add Reason header field
>>
>>        The SIP entity adds one or more Reason header fields to the hi-
>> ------^
>> change "The" to "If the response code is not 1xx, the"
>>        targeted-to-uri in the (newly) cached hi-entry reflecting the SIP
>>        response code in the non-100 response, per the procedures of
>> ---------------^^^^^^^^^^^^^^^^^^^^^^^^^^^^
>> delete(?)
>>        Section 10.2.
>>
>> This change results in:
>>
>>        If the response code is not 1xx, the SIP entity adds one or more
>>        Reason header fields to the hi-targeted-to-uri in the (newly)
>>        cached hi-entry reflecting the SIP response, per the procedures
>>        of Section 10.2.
>>
>> [MB] Note that the section is specific to non-100 responses, which is
>> why it is written as it currently is.
>> I don't see that your suggested text adds any more information - i.e.,
>> the IF isn't necessary.  We are stating specific behavior for that
>> specific step and an "IF" isn't necessary.  I do not think this change
>> is necessary. [/MB]
>>
>>     10.2.  Reason in the History-Info Header Field
>>
>>     A Reason header field is added to the "headers" component in an hi-
>>     targeted-to-uri when the hi-entry is added to the cache based upon
>>     the receipt of a non-100 or non-2xx SIP response, as described in
>>     Section 9.3.  If the Reason header field is being added due to
>>     receipt of an explicit SIP response and the response contains any
>>     Reason header fields (see [RFC3326]), then the SIP entity MUST
>>     include the Reason header fields in the "headers" component of the
>>     hi-targeted-to-uri in the last hi-entry added to the cache, unless
>>     the hi-targeted-to-uri is a Tel-URI.  If the SIP response does not
>>     contain a Reason header field, the SIP entity MUST include a Reason
>>     header field, containing the SIP Response Code, in the "headers"
>>     component of the hi-targeted-to-uri in the last hi-entry added to the
>>     cache, unless the hi-targeted-to-uri is a Tel-URI.
>>
>> The changes to this paragraph are extensive enough that I am not
>> showing the individual differences.  Note I have changed "the last
>> hi-entry added to the cache" to "the corresponding hi-entry".  The
>> latter makes more sense to me, and specifies the same hi-entry.  But
>> others may prefer the current description, in which case we can retain
>> it.
>>
>> The new version of this paragraph is:
>>
>>     A Reason header field is added to the "headers" component in an hi-
>>     targeted-to-uri when a non-100, non-2xx SIP response is received to
>>     the corresponding INVITE, or when a response is received containing
>>     the hi-entry with a Reason header value, as described in Section
>>     9.3.
>> [MB] You have dropped the reference to the cache here and that very
>> important.  I think it's obvious that we
>> are dealing with the corresponding INVITE.  When we talk about SIP
>> responses, we do not qualify it as "the SIP response received to the
>> corresponding INVITE".  But, it certainly causes no harm.  I would
>> suggest that sentence read:
>>     A Reason header field is added to the "headers" component in an hi-
>>     targeted-to-uri when the hi-entry is added to the cache based upon
>>     the receipt of a non-100 or non-2xx SIP response to the
>> corresponding INVITE, as described in
>>     Section 9.3.
>>
>> [/MB]
>>
>>     If the Reason header value is being added to an hi-targeted-to-uri
>>     due to receipt of an explicit SIP response to the corresponding
>>     INVITE and the response contains any Reason header fields (see
>>     [RFC3326]), then the SIP entity MUST also include the Reason header
>>     fields in the "headers" component of the hi-targeted-to-uri in the
>>     corresponding cached hi-entry, unless the hi-targeted-to-uri is a
>>    ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
>>     Tel-URI.  If the SIP response does not contain a Reason header
>>     field with a protocol of "SIP", the SIP entity MUST include a
>>     Reason header field, containing the SIP Response Code, in the
>>     "headers" component of the hi-targeted-to-uri in the corresponding
>>     cached hi-entry, unless the hi-targeted-to-uri is a Tel-URI.
>>
>> [MB]  Note, you dropped "field" from the "Reason header value above,
>> which is not correct. So the real changes I see above are with regards
>> to highlighting that it's the corresponding cached hi-entry (rather
>> than the last.  I have annotated above that that is the only change
>> (AFAICT).  I can make that change as follows:
>> OLD:
>>      If the Reason header field is being added due to receipt of an
>> explicit
>>      SIP response and the response contains any
>>      Reason header fields (see [RFC3326]), then the SIP entity MUST
>>      include the Reason header fields in the "headers" component of the
>>      hi-targeted-to-uri in the last hi-entry added to the cache, unless
>>      the hi-targeted-to-uri is a Tel-URI.
>>
>> NEW:
>>     If the Reason header field is being added due to receipt of an
>> explicit
>>     SIP response to the corresponding INVITE
>>                                      ^^^^^^^^^^^^^^^^^^^^^^
>>     and the response contains any
>>     Reason header fields (see [RFC3326]), then the SIP entity MUST
>>     include the Reason header fields in the "headers" component of the
>>     hi-targeted-to-uri in the corresponding cached hi-entry,  unless
>>                                           ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
>>     the hi-targeted-to-uri is a Tel-URI.
>> [/MB]
>>
>> There is a slight change here in that if the response contains
>> "Reason: Q.850;cause=14", then both the Q.850 response code and the
>> SIP response code will be recorded in the cached hi-entry.  The
>> previous text caused only the Q.850 response code to be recorded.
>> [MB] I don't see that the previous text said that only the Q.850
>> response code was recorded.
>> The text says that "Reason header fields" as in plural fields are added.
>> [/MB]
>>
>> If a Reason header includes a reason-value with protocol "SIP", that
>> reason-value is recorded instead of the response code in the response.
>> (This is the same as the previous text.)
>>
>> ==========
>>
>>> 2. During the life of the document, there was confusion about how
>>> hi-entries were to be organized.  The intention of the original writer
>>> and the current draft is that hi-entries are accumulated in a cache in
>>> the order the entity receives them (in requests and responses),
>>> allowing hi-entries that have the *same* index but *different* URIs,
>>> and (implicitly) eliminating duplicates.
>>
>> [MB] It wasn't just the original author and the current author, this
>> has been the intent of
>> the document and the approach was agreed by the WG. Certainly, there were
>> some
>> errors that might have lead one to believe otherwise, but that was
>> never the intent. [/MB]
>>>
>>>
>>> But some people who have worked on the draft (including myself) were
>>> for a while mistaken in believing that the hi-entries were to be
>>> organized into a tree, and hence the index values needed to be unique,
>>> and the History-Info header would list the hi-entries in tree-walking
>>> order (technically, "preorder").  The confusion was abetted by there
>>> being no examples that distinguish the two techniques.
>>
>> [MB] I think the confusion is that if everyone supported HI, you would
>> get a tree.
>> You can still get a somewhat pruned tree using the indices.  I think
>> the confusion is that the RFC never
>> had any intention of defining application behavior.  It is merely
>> recording what happens at the SIP protocol level. That's it.  How an
>> application uses the information (whether to build a tree or whatever)
>> is out of scope. [/MB]
>>>
>>> Unfortunately, the -11 draft doesn't eliminate all the remnants of the
>>> mistake.  I haven't gone through -11 in detail, but "preorder" appears
>>> in section 9.2 and section 10.3, so at least those sections describe
>>> the accumulation of hi-entries incorrectly.
>>
>>
>> The fix to this issue overlaps the fix for issue #4:
>>
>>> 4. If a downstream entity that does not support History-Info forks a
>>> request to entities that do support History-Info, an entity can
>>> receive two hi-entries with the same index that were not generated for
>>> the same fork of the request.  Conversely, an entity will receive
>>> responses that contain hi-entries that are duplicates of entries that
>>> are already in the cache.
>>>
>>> These facts imply that there is a process by which an entity
>>> determines whether a received hi-entry is "the same" as an hi-entry
>>> already in the cache (and so doesn't need to be added again).  Since
>>> index values cannot be depended upon to be unique, we really ought to
>>> state what the comparison rule is.
>>>
>>> I think it is sufficient to use the rule "the same index value and the
>>> URIs are equivalent (per RFC 3261 section 19.1.4, but ignoring the
>>> Reason header fields)".  This won't always give the result one would
>>> want (if a non-supporting downstream element forks the request to two
>>> target URIs that ultimately arrive at the same element, two hi-entries
>>> could be generated with the same index and URI), but this rule is
>>> simple, well-defined, and should work well in almost all situations.
>>
>>
>> In addition, when a response is received, it may contain hi-entries
>> that are duplicates of hi-entries already in the cache, except they
>> contain Reason header values that are not in the cached hi-entries.  I
>> believe we intend that the cached hi-entries be updated to include the
>> received Reason header values.
>>
>> The edits to fix these three issues are:
>>
>>     9.2.  Sending a Request with History-Info
>>
>>     When sending a request, a SIP entity MUST include all the hi-entries
>>     from the cache that was created per Section 9.1.  In addition, the
>>     SIP entity MUST add a new hi-entry to the outgoing request, but the
>>     SIP entity MUST NOT add the hi-entry to the cache at this time.  The
>> --------------------------------------------------------------------^
>> Change this sentence to:  "The hi-entries in the outgoing request's
>> History-Info header are in the order of the hi-entries in the cache,
>> which is the order in which they were added to the cache."
>>
>> [MB] That's fine. And, just for clarification, the statement below is
>> the OLD and the one above is the NEW [/MB]
>>
>>     hi-entries in the outgoing request's History-Info header field is the
>>     preorder of the tree of hi-entries, that is, by the lexicographic
>>     ordering of the hi-indexes.  The new hi-entry is populated as
>>     follows:
>>
>>     9.3.  Receiving a Response with History-Info or Request Timeouts
>>
>>     Step 1:  Add hi-entry to cache
>>
>>        The SIP entity MUST add the hi-entry that was added to the request
>> -----------------------------^
>> For clarity insert "to the cache" here, and...
>>        that received the non-100 response or timed out to the cache, if
>> ------------------------------------------------------^^^^^^^^^^^^
>> ... remove "to the cache" here.
>>        it was not already cached.  The hi-entry MUST be added to the
>> ----------------------------------^
>> Change this sentence to "If so, the hi-entry MUST be added to the end
>> of the cache."
>>
>>        cache in ascending order as indicated by the values in the hi-
>>        index parameters of the hi-entries (e.g., 1.2.1 comes after 1.2
>>        but before 1.2.2 or 1.3).
>>
>> [MB] I can't decipher the changes you are suggesting above.  I think I
>> see your point.  However,
>> there is something very subtle here.  This is dealing with responses
>> and we add the hi-entries once we
>> get the response.  However, the value of the hi-index actually does
>> reflect a chronological order in terms of generating requests.  As
>> discussed previously, you lose that chronological ordering across
>> entities, but for a specific
>> SIP entity they do reflect the chronological processing at that
>> entity.  So, I don't think this is broken.  With the cache
>> model, we aren't adding them to the cache until we get the response
>> even though they were added to the request (in both chronological and
>> lexicographical order).
>> [/MB]
>>
>>
>>     9.3.  Receiving a Response with History-Info or Request Timeouts
>>
>>     Step 3:  Add additional hi-entries
>>
>>        The SIP entity MUST also add to the cache any hi-entries received
>>        in the response that are not already in the cache.  This situation
>>        can occur when the entity that generated the non-100 response
>>        retargeted the request before generating the response.  As per
>>        Step 1, the hi-entries MUST be added to the cache in ascending
>>        order as indicated by the values in the hi-index parameters of the
>>        hi-entries
>>
>> This paragraph needs more extensive changes:
>> [MB] It's not clear to me why you think it needs to be done this way.
>> This is still mapping back to the original request sent by a specific
>> entity (e.g., Proxy A).   I think you are confused by the scenario
>> being described.  The retargeting was not done by Proxy A that is
>> caching the hi-entries.  It was done by Proxy B that received the
>> request and it retargeted before returning the 1xx response thus there
>> is an additional hi-entry that Proxy A should add to its cache. [/MB]
>>
>>     Step 3:  Add and update additional hi-entries
>>
>>        The SIP entity MUST also add to the cache any hi-entries received
>>        in the response that are not already in the cache.  This
>>        can occur when the entity that generated the non-100 response
>>        retargeted the request before generating the response.  Such
>>        hi-entries MUST be added to the end of the cache in the order
>>        they appear in the History-Info header field of the response.
>>
>>        If an hi-entry in the response contains Reason header value(s)
>>        and the matching hi-entry in the cache does not, the Reason
>>        header value(s) MUST be added to the cached hi-entry.  This can
>>        occur when the hi-entry was added to the cache by a provisional
>>        response to an INVITE that was further retargeted, and later a
>>        final response to that INVITE is received.
>>
>>        In these cases, an hi-entry in a response's History-Info header
>>        field and an hi-entry in the cache are considered matching when
>>        they have the same index value and the hi-targeted-to-uris are
>>        equivalent (per RFC 3261 section 19.1.4, but ignoring the Reason
>>        header fields).
>>
>>     10.2.  Reason in the History-Info Header Field
>>
>>     A Reason header field is added to the "headers" component in an hi-
>>     targeted-to-uri when the hi-entry is added to the cache based upon
>> ----------------------------------------^^^^^^^^
>> change to "added to, or updated within,"
>>
>>     the receipt of a non-100 or non-2xx SIP response, as described in
>>     Section 9.3.  [...]
>> [MB] Why "updated within"?  [/MB]
>>
>>     10.3.  Indexing in the History-Info Header Field
>>
>>     In order to maintain ordering and accurately reflect the retargeting
>>     of the request, the SIP entity MUST add a hi-index to each hi-entry.
>>     Per the syntax in Section 5, the hi-index consists of a series of
>>     nonnegative integer separated by dots (e.g., 1.1.2).  Each dot
>>     reflects a SIP forwarding hop.  The nonnegative integer following
>>     each dot reflects the order in which a request was retargeted at the
>>     hop.  The highest nonnegative integer at each hop reflects the number
>>     of entities to which the request has been retargeted at the specific
>>     hop (i.e., the number of branches) at the time that the request
>>     represented by this hi-entry was generated.  Thus, the indexing
>>     results in a logical tree representation for the history of the
>>     request and the hi-entries are given in the preorder of the tree.
>> -----------^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
>> delete this clause
>> [MB] Okay. [/MB]
>>
>> ==========
>>
>>> 3. Due to the confusion discussed in #2, we have to consider whether
>>> consensus was actually found in the WGLC.
>>
>> [MB] We had consensus in WGLC.  Your comments came well after WG last call
>> and after IETF LC.
>>
>>> We thought we had
>>> consensus, but people did not have the same understanding of this
>>> aspect of the draft.  (Ugh.)
>>
>>
>> This is a procedural issue, not a technical one.  I don't see any
>> objection to correcting the normative text if everyone who cares
>> agrees that the changes express what we intended.  (We should check
>> this point with the AD.)
>>
>> [MB] I do not see that we are changing any of the intended normative
>> behavior but rather fixing things that didn't quite reflect the
>> normative behavior as stated elsewhere (i.e., inconsistencies). [/MB]
>>
>> ==========
>>
>>> 5. In draft-ietf-sipcore-rfc4244bis-callflows-01, examples 3.6 and 3.7
>>> do not show the History-Info value in the final 200 response to the
>>> UAC.  This final value would be useful, in particular, because they
>>> give further examples of how proxies process History-Info.
>>
>>
>> I was mistaken here:  The final 200 response from the example.com
>> proxy to the UAC in these examples has the same History-Info as the
>> 200 response arriving at the example.com proxy, because there is no
>> History-Info information that the proxy knows that is not known to the
>> UAS that generated the 200 response.  Thus there is nothing
>> interesting to document.
>>
>> ==========
>>
>>> 6. Interestingly, despite the fact that there can be duplicate index
>>> values, when a request is received by a UAS, there is no ambiguity in
>>> the history when tracing *back* from the last hi-entry (the one that
>>> denotes the UAS).  This is because even if this UAS is "down one fork"
>>> from a forking non-supporting proxy, none of the hi-entries generated
>>> "down the other fork" will be present in the request the UAS sees.
>>>
>>> This is in contrast to the situation at the UAC, which may see
>>> hi-entries with duplicate indexes, and the interpretation of "rc",
>>> "mp", and "np" parameters can be ambiguous.
>>
>>
>> This is an interesting observation, but it does not require
>> documentation in the RFCs.
>>
>> ==========
>>
>>> 7. In example 3.6 message F6 in
>>> draft-ietf-sipcore-rfc4244bis-callflows-01, the History-Info is:
>>>
>>>     History-Info: <sip:bob@example.com>;index=1
>>>     History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302>;\
>>>                        index=1.1;rc=1
>>>     History-Info: <sip:carol@example.com>;index=1.2;mp=1
>>>     History-Info: <sip:carol@192.0.2.4?Reason=SIP%3Bcause%3D408>;\
>>>                        index=1.2.1;rc=1.2
>>>     History-Info: <sip:vm@example.com;\
>>>                        target=sip:bob%40example.com;cause=408>;\
>>>                        index=1.3;mp=1.2
>>>     History-Info: <sip:vm@192.0.2.6;\
>>>                        target=sip:bob%40example.com;cause=408>;\
>>>                        index=1.3.1;rc=1.3
>>>
>>> This is what we intend, as it shows both where the index=1.3 URI came
>>> from (sip:bob@example.com) as well as why the index=1.3 fork was
>>> generated (because sip:carol@example.com failed).  But it points out
>>> that we need to improve the description in rfc4244bis section 5 of
>>> what the values of the "rc" (etc.) parameters are.  A naive reader
>>> might think "rc=1.2" in the 1.3 entry means that
>>> <sip:vm@example.com;target=sip:bob%40example.com;cause=408> was
>>> generated as a contact for <sip:carol@192.0.2.4>.  Instead, it means
>>> that the 1.3 fork was generated *because* the 1.2 fork returned a
>>> response; it was generated *from* the index=1 URI.
>>>
>>> Actually, the real rules for interpreting "rc" (etc.) are not simple:
>>
>> [MB] The above is an example, thus the description is not complete. [/MB]
>>>
>>>
>>> 1) If the index-val in the param points to an entry with a Reason
>>> header-param whose value is "SIP;cause=3xx" (after removing
>>> escaping!), then:
>>> - the URI was one of the Contact values in the 3xx response
>>> - that response was why the new fork was generated.
>>>
>>> 2) If the index-val in the param points to the parent of this
>>> hi-entry, then:
>>> - the URI was generated from the URI of the parent hi-entry
>>> - the fork was not generated due to the response from any other fork.
>>>
>>> 3) If the index-val in the param points to an hi-entry that is not the
>>> parent, and that hi-entry does not contain a Reason header-param
>>> containing a 3xx status, then:
>>> - the URI was generated from the URI of the parent hi-entry
>>> - the fork was generated due to the response from the hi-entry with
>>> the specified index-val.
>>>
>>> And this doesn't match the section 5 definition of the value of "rc"
>>> (etc.), which is:
>>>
>>>           The "rc" header field parameter contains the value of the
>>>           hi-index in the hi-entry with an hi-targeted-to-uri that
>>>           reflects the Request-URI that was retargeted
>>
>>
>> In section 5, these paragraphs need to be revised:
>>
>>     o  hi-target-param: An optional parameter reflecting the mechanism by
>>        which the Request URI captured in the hi-targeted-to-uri in the
>>        History-Info header field value (hi-entry) was determined.  This
>>        parameter is either an "rc", "mp" or "np" header field parameter,
>>        which is interpreted as follows:
>>
>>           "rc": The hi-targeted-to-URI represents a change in Request-URI
>>           while the target user remains the same.  This occurs for
>>           example when user has multiple AoRs as an alias.  The "rc"
>>           header field parameter contains the value of the hi-index in
>>           the hi-entry with an hi-targeted-to-uri that reflects the
>>           Request-URI that was retargeted
>>
>>           "mp": The hi-targeted-to-URI represents a user other than the
>>           target user associated with the Request-URI in the incoming
>>           request that was retargeted.  This occurs when a request is
>>           statically or dynamically retargeted to another user
>>           represented by an AoR unassociated with the AoR of the original
>>           target user.  The "mp" header field parameter contains the
>>           value of the hi-index in the hi-entry with an hi-targeted-to-
>>           uri that reflects the Request-URI that was retargeted, thus
>>           identifying the "mapped from" target.
>>
>>           "np": The hi-targeted-to-URI represents that there was no
>>           change in Request-URI.  This would apply for example when a
>>           proxy merely forwards a request to a next hop proxy and loose
>>           routing is used.  The "np" header field parameter contains the
>>           value of the hi-index in the hi-entry with an hi-targeted-to-
>>           uri that reflects the Request-URI that was copied unchanged
>>           into the request represented by this hi-entry.  That value will
>>           usually be the hi-index of the parent hi-entry of this hi-
>>           entry.
>>
>> The revised paragraphs are:
>>
>> [MB] I don't agree with this proposal - that's way too much detail for
>> this section.  I believe that the normative
>> sections describe how/when you add the "rc" tag which should match
>> what you are describing in your new bullets.    I will note that this
>> section has been updated several times both adding more text and
>> reducing the text in this section.  What you have proposed for the
>> bullets below (i.e., the "rc", "mp" etc.) is some of the level of
>> detail we had previously (in the -02).   I believe the text you
>> suggest is application logic - note that items 2 & 4 describe the "rc"
>> scenarios and this wasn't intended to be comprehensive.  If you can
>> highlight what's missing in those 2 bullets, we can perhaps make some
>> changes there.   I will note if these suggestions had been made much
>> earlier in the process, it would have been much more helpful - much of
>> the text you mention has been in the document since around the -02.
>> We did several rounds of major edits while developing this document
>> both while it was an individual and as a WG document.  Note that this
>> document became a WG document literally 3 years ago.  It's quite
>> disruptive to be raising this issues at this stage. [/MB]
>>
>>     [This first paragraph is unchanged:]
>>     o  hi-target-param: An optional parameter reflecting the mechanism by
>>        which the Request URI captured in the hi-targeted-to-uri in the
>>        History-Info header field value (hi-entry) was determined.  This
>>        parameter is either an "rc", "mp" or "np" header field parameter,
>>        which is interpreted as follows:
>>
>>           "rc": The hi-targeted-to-URI represents a change in Request-URI
>>           while the target user remains the same.  This occurs for
>>           example when user has multiple AoRs as an alias or when
>>           forwarding the request to a registered contact of an AoR.
>>
>>           "mp": The hi-targeted-to-URI represents a user other than the
>>           target user associated with the Request-URI in the incoming
>>           request that was retargeted.  This occurs when a request is
>>           statically or dynamically retargeted to another user
>>           represented by an AoR unassociated with the AoR of the original
>>           target user.
>>
>>           "np": The hi-targeted-to-URI represents that there was no
>>           change in Request-URI.  This would apply for example when a
>>           proxy merely forwards a request to a next hop proxy and loose
>>           routing is used.  The "np" header field parameter contains the
>>           value of the hi-index in the parent hi-entry of this hi-entry.
>>
>>        If the index-val in the parameter points to the parent hi-entry of
>> this
>>        hi-entry, then:
>>        - the URI was generated from the URI of the parent hi-entry
>>        - the fork was not generated due to the response from any other
>> fork.
>>
>>        If the index-val in the parameter points to an hi-entry with a
>> Reason
>>        header value whose protocol is "SIP" and whose cause begins with
>> "3",
>>        then:
>>        - the URI was one of the Contact values in the 3xx response
>>        - the fork was generated due to the 3xx response from the hi-entry
>> with
>>        the specified index-val.
>>
>>        If the index-val in the param points to an hi-entry that is not the
>>        parent, and that hi-entry does not contain a Reason header value
>>        containing a SIP 3xx cause, then:
>>        - the URI was generated from the URI of the parent hi-entry
>>        - the fork was generated due to the response from the hi-entry with
>>        the specified index-val.
>>
>> ======================================================================
>>
>> I believe that is everything to be done.
>>
>> Dale
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore

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Subject: Re: [sipcore] Fwd: Updates to draft-ietf-sipcore-rfc4244bis-11, version 1.
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On Tue, Feb 19, 2013 at 10:50 AM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
> Mary & Dale,
>
> This is very hard to follow.
> But I do get it that the two of you are getting closer but still not in
> total agreement. Can the two of you reach a mutually satisfactory
> conclusion?
[MB] In the end, this is a WG, chair and AD decision since this
document has completed IETF LC. IMHO, the bar is much higher for
changes at this stage in the process. If there is a lone voice with
concerns that are not agreed by the WG and for which the justification
is not extremely compelling, then I do not believe the changes should
be made.  As I said in my response, these changes could have been much
more easily incorporated earlier in the process.  For example, the
last set of proposed changes are around text that has been heavily
edited as well as a proposal to state things that are already in the
document in a different manner. We can always find a way to edit text
in our documents - at some point we say we're done which is usually
after we've completed IETF LC unless the IESG review raises critical
issues.  [/MB]
>
> I'd love to have a new version with a direct diff to evaluate it all.
[MB] I can generate a new version to reflect the changes that have
been agreed. [/MB]
>
> I must say that I am surprised that so much effort was put into the
> hierarchical numbering scheme for entries and yet the conclusion is that the
> entries are not sorted in accord with this hierarchy. (Or did I get that
> wrong?)
>
>         Thanks,
>         Paul
>
>
> On 2/19/13 11:16 AM, Mary Barnes wrote:
>>
>> Hi all,
>>
>> This is a set of suggested changes from Dale based upon the comments
>> he sent to the list previously:
>> http://www.ietf.org/mail-archive/web/sipcore/current/msg05489.html
>>
>> Responses are inline below [MB].   Based upon my responses, I do not
>> believe any of the changes impact normative behavior (as intended)
>> thus another WG and/or IETF LC should not be necessary, but I will
>> leave that decision to the chairs/AD.
>>
>> Mary.
>>
>>
>> ---------- Forwarded message ----------
>> From: Dale R. Worley <worley@ariadne.com>
>> Date: Wed, Feb 6, 2013 at 9:09 AM
>> Subject: Updates to draft-ietf-sipcore-rfc4244bis-11, version 1.
>> To: mary.h.barnes@gmail.com, pkyzivat@alum.mit.edu
>>
>>
>> Here are my proposed fixes to the -11 version.  I believe that all of
>> these changes are non-controverisial, in that they adjust the document
>> to express what we all agree it ought to mean.  But in a number of
>> places, the prescribed procedures are actually different from the
>> current text.  Please review this and get back to us with your
>> thoughts.
>>
>> Thanks,
>>
>> Dale
>>
>> ======================================================================
>>
>> Updates to draft-ietf-sipcore-rfc4244bis-11, version 1.
>>
>> This document is organized as a series of responses to the points in
>> my e-mail of 18 Jan
>> (http://www.ietf.org/mail-archive/web/sipcore/current/msg05489.html).
>>
>> ==========
>>
>>> From: worley@ariadne.com (Dale R. Worley)
>>> To: SIPCORE <sipcore@ietf.org>
>>> Subject: [sipcore] Questions about draft-ietf-sipcore-rfc4244bis-11.txt
>>> Date: Fri, 18 Jan 2013 12:10:40 -0500 (EST)
>>>
>>> I've got a few questions about 4244bis.  I think that there are
>>> understood answers to most of them, but we might have to update the
>>> wording to remove ambiguity.
>>
>>
>> ==========
>>
>>> 1. When an entity receives a (non-100) provisional response to a
>>> request, should it modify the hi-entry by adding a Reason value
>>> carrying the status of the response?  The current text (section 10.2)
>>> clearly requires it.  If the request eventually receives a failure
>>> response, the 1xx Reason will (probably) later be replaced by the
>>> failure code.  (But not necessarily:  see section 10.2 paragraph 3.)
>>>
>>> The only example in draft-ietf-sipcore-rfc4244bis-callflows-01 of this
>>> situation is section 3.1 message F8, and the 180 status is not
>>> reported in a Reason header-value for <sip:office@example.com> (which
>>> thus contradicts section 10.2).
>>>
>>> The alternative is that 1xx responses do not set the Reason value
>>> (although they do add to the cache any new hi-entries they carry).
>>> This would require changing the text in section 9.3 step 2 and section
>>> 10.2.
>>
>>
>> NOTE:  I am assuming that non-100 provisional responses are intended
>> to carry hi-entries upstream, and that proxies that see those
>> responses are intended to cache the hi-entries carried in them.
>> (Those cached hi-entries can later be updated by a final response
>> which adds a Reason header value to them.)  If this is not so, these
>> changes need to be revised.
>>
>> As far as I can tell, we have never intended for receipt of a
>> provisional (1xx) response to an INVITE to set a Reason header on the
>> corresponding cached hi-entry.  The only example in 4244bis-callflows
>> of this situation is message F6 of section 3.1, in which there would
>> be a Reason header on hi-entries 1.2 and 1.2.1 -- and there is not.
>> And RFC 4244, which also has the Reason header value mechanism, does
>> not specify that 1xx responses set Reason header values.
>>
>> So I propose that we all agree that 1xx responses should not generate
>> Reason headers and we edit the text accordingly.
>> [MB] Correct.  That's been the intent for both 4244 and 4244bis.
>> It's mentioned in 9.4 that the HI is not sent in 100 responses (since
>> those do not result in retargeting), thus there would be no reason to
>> add a Reason header to anything.  [/MB]
>>
>>   The changes are:
>>
>>     section 9.3 step 2
>>
>>     Step 2:  Add Reason header field
>>
>>        The SIP entity adds one or more Reason header fields to the hi-
>> ------^
>> change "The" to "If the response code is not 1xx, the"
>>        targeted-to-uri in the (newly) cached hi-entry reflecting the SIP
>>        response code in the non-100 response, per the procedures of
>> ---------------^^^^^^^^^^^^^^^^^^^^^^^^^^^^
>> delete(?)
>>        Section 10.2.
>>
>> This change results in:
>>
>>        If the response code is not 1xx, the SIP entity adds one or more
>>        Reason header fields to the hi-targeted-to-uri in the (newly)
>>        cached hi-entry reflecting the SIP response, per the procedures
>>        of Section 10.2.
>>
>> [MB] Note that the section is specific to non-100 responses, which is
>> why it is written as it currently is.
>> I don't see that your suggested text adds any more information - i.e.,
>> the IF isn't necessary.  We are stating specific behavior for that
>> specific step and an "IF" isn't necessary.  I do not think this change
>> is necessary. [/MB]
>>
>>     10.2.  Reason in the History-Info Header Field
>>
>>     A Reason header field is added to the "headers" component in an hi-
>>     targeted-to-uri when the hi-entry is added to the cache based upon
>>     the receipt of a non-100 or non-2xx SIP response, as described in
>>     Section 9.3.  If the Reason header field is being added due to
>>     receipt of an explicit SIP response and the response contains any
>>     Reason header fields (see [RFC3326]), then the SIP entity MUST
>>     include the Reason header fields in the "headers" component of the
>>     hi-targeted-to-uri in the last hi-entry added to the cache, unless
>>     the hi-targeted-to-uri is a Tel-URI.  If the SIP response does not
>>     contain a Reason header field, the SIP entity MUST include a Reason
>>     header field, containing the SIP Response Code, in the "headers"
>>     component of the hi-targeted-to-uri in the last hi-entry added to the
>>     cache, unless the hi-targeted-to-uri is a Tel-URI.
>>
>> The changes to this paragraph are extensive enough that I am not
>> showing the individual differences.  Note I have changed "the last
>> hi-entry added to the cache" to "the corresponding hi-entry".  The
>> latter makes more sense to me, and specifies the same hi-entry.  But
>> others may prefer the current description, in which case we can retain
>> it.
>>
>> The new version of this paragraph is:
>>
>>     A Reason header field is added to the "headers" component in an hi-
>>     targeted-to-uri when a non-100, non-2xx SIP response is received to
>>     the corresponding INVITE, or when a response is received containing
>>     the hi-entry with a Reason header value, as described in Section
>>     9.3.
>> [MB] You have dropped the reference to the cache here and that very
>> important.  I think it's obvious that we
>> are dealing with the corresponding INVITE.  When we talk about SIP
>> responses, we do not qualify it as "the SIP response received to the
>> corresponding INVITE".  But, it certainly causes no harm.  I would
>> suggest that sentence read:
>>     A Reason header field is added to the "headers" component in an hi-
>>     targeted-to-uri when the hi-entry is added to the cache based upon
>>     the receipt of a non-100 or non-2xx SIP response to the
>> corresponding INVITE, as described in
>>     Section 9.3.
>>
>> [/MB]
>>
>>     If the Reason header value is being added to an hi-targeted-to-uri
>>     due to receipt of an explicit SIP response to the corresponding
>>     INVITE and the response contains any Reason header fields (see
>>     [RFC3326]), then the SIP entity MUST also include the Reason header
>>     fields in the "headers" component of the hi-targeted-to-uri in the
>>     corresponding cached hi-entry, unless the hi-targeted-to-uri is a
>>    ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
>>     Tel-URI.  If the SIP response does not contain a Reason header
>>     field with a protocol of "SIP", the SIP entity MUST include a
>>     Reason header field, containing the SIP Response Code, in the
>>     "headers" component of the hi-targeted-to-uri in the corresponding
>>     cached hi-entry, unless the hi-targeted-to-uri is a Tel-URI.
>>
>> [MB]  Note, you dropped "field" from the "Reason header value above,
>> which is not correct. So the real changes I see above are with regards
>> to highlighting that it's the corresponding cached hi-entry (rather
>> than the last.  I have annotated above that that is the only change
>> (AFAICT).  I can make that change as follows:
>> OLD:
>>      If the Reason header field is being added due to receipt of an
>> explicit
>>      SIP response and the response contains any
>>      Reason header fields (see [RFC3326]), then the SIP entity MUST
>>      include the Reason header fields in the "headers" component of the
>>      hi-targeted-to-uri in the last hi-entry added to the cache, unless
>>      the hi-targeted-to-uri is a Tel-URI.
>>
>> NEW:
>>     If the Reason header field is being added due to receipt of an
>> explicit
>>     SIP response to the corresponding INVITE
>>                                      ^^^^^^^^^^^^^^^^^^^^^^
>>     and the response contains any
>>     Reason header fields (see [RFC3326]), then the SIP entity MUST
>>     include the Reason header fields in the "headers" component of the
>>     hi-targeted-to-uri in the corresponding cached hi-entry,  unless
>>                                           ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
>>     the hi-targeted-to-uri is a Tel-URI.
>> [/MB]
>>
>> There is a slight change here in that if the response contains
>> "Reason: Q.850;cause=14", then both the Q.850 response code and the
>> SIP response code will be recorded in the cached hi-entry.  The
>> previous text caused only the Q.850 response code to be recorded.
>> [MB] I don't see that the previous text said that only the Q.850
>> response code was recorded.
>> The text says that "Reason header fields" as in plural fields are added.
>> [/MB]
>>
>> If a Reason header includes a reason-value with protocol "SIP", that
>> reason-value is recorded instead of the response code in the response.
>> (This is the same as the previous text.)
>>
>> ==========
>>
>>> 2. During the life of the document, there was confusion about how
>>> hi-entries were to be organized.  The intention of the original writer
>>> and the current draft is that hi-entries are accumulated in a cache in
>>> the order the entity receives them (in requests and responses),
>>> allowing hi-entries that have the *same* index but *different* URIs,
>>> and (implicitly) eliminating duplicates.
>>
>> [MB] It wasn't just the original author and the current author, this
>> has been the intent of
>> the document and the approach was agreed by the WG. Certainly, there were
>> some
>> errors that might have lead one to believe otherwise, but that was
>> never the intent. [/MB]
>>>
>>>
>>> But some people who have worked on the draft (including myself) were
>>> for a while mistaken in believing that the hi-entries were to be
>>> organized into a tree, and hence the index values needed to be unique,
>>> and the History-Info header would list the hi-entries in tree-walking
>>> order (technically, "preorder").  The confusion was abetted by there
>>> being no examples that distinguish the two techniques.
>>
>> [MB] I think the confusion is that if everyone supported HI, you would
>> get a tree.
>> You can still get a somewhat pruned tree using the indices.  I think
>> the confusion is that the RFC never
>> had any intention of defining application behavior.  It is merely
>> recording what happens at the SIP protocol level. That's it.  How an
>> application uses the information (whether to build a tree or whatever)
>> is out of scope. [/MB]
>>>
>>> Unfortunately, the -11 draft doesn't eliminate all the remnants of the
>>> mistake.  I haven't gone through -11 in detail, but "preorder" appears
>>> in section 9.2 and section 10.3, so at least those sections describe
>>> the accumulation of hi-entries incorrectly.
>>
>>
>> The fix to this issue overlaps the fix for issue #4:
>>
>>> 4. If a downstream entity that does not support History-Info forks a
>>> request to entities that do support History-Info, an entity can
>>> receive two hi-entries with the same index that were not generated for
>>> the same fork of the request.  Conversely, an entity will receive
>>> responses that contain hi-entries that are duplicates of entries that
>>> are already in the cache.
>>>
>>> These facts imply that there is a process by which an entity
>>> determines whether a received hi-entry is "the same" as an hi-entry
>>> already in the cache (and so doesn't need to be added again).  Since
>>> index values cannot be depended upon to be unique, we really ought to
>>> state what the comparison rule is.
>>>
>>> I think it is sufficient to use the rule "the same index value and the
>>> URIs are equivalent (per RFC 3261 section 19.1.4, but ignoring the
>>> Reason header fields)".  This won't always give the result one would
>>> want (if a non-supporting downstream element forks the request to two
>>> target URIs that ultimately arrive at the same element, two hi-entries
>>> could be generated with the same index and URI), but this rule is
>>> simple, well-defined, and should work well in almost all situations.
>>
>>
>> In addition, when a response is received, it may contain hi-entries
>> that are duplicates of hi-entries already in the cache, except they
>> contain Reason header values that are not in the cached hi-entries.  I
>> believe we intend that the cached hi-entries be updated to include the
>> received Reason header values.
>>
>> The edits to fix these three issues are:
>>
>>     9.2.  Sending a Request with History-Info
>>
>>     When sending a request, a SIP entity MUST include all the hi-entries
>>     from the cache that was created per Section 9.1.  In addition, the
>>     SIP entity MUST add a new hi-entry to the outgoing request, but the
>>     SIP entity MUST NOT add the hi-entry to the cache at this time.  The
>> --------------------------------------------------------------------^
>> Change this sentence to:  "The hi-entries in the outgoing request's
>> History-Info header are in the order of the hi-entries in the cache,
>> which is the order in which they were added to the cache."
>>
>> [MB] That's fine. And, just for clarification, the statement below is
>> the OLD and the one above is the NEW [/MB]
>>
>>     hi-entries in the outgoing request's History-Info header field is the
>>     preorder of the tree of hi-entries, that is, by the lexicographic
>>     ordering of the hi-indexes.  The new hi-entry is populated as
>>     follows:
>>
>>     9.3.  Receiving a Response with History-Info or Request Timeouts
>>
>>     Step 1:  Add hi-entry to cache
>>
>>        The SIP entity MUST add the hi-entry that was added to the request
>> -----------------------------^
>> For clarity insert "to the cache" here, and...
>>        that received the non-100 response or timed out to the cache, if
>> ------------------------------------------------------^^^^^^^^^^^^
>> ... remove "to the cache" here.
>>        it was not already cached.  The hi-entry MUST be added to the
>> ----------------------------------^
>> Change this sentence to "If so, the hi-entry MUST be added to the end
>> of the cache."
>>
>>        cache in ascending order as indicated by the values in the hi-
>>        index parameters of the hi-entries (e.g., 1.2.1 comes after 1.2
>>        but before 1.2.2 or 1.3).
>>
>> [MB] I can't decipher the changes you are suggesting above.  I think I
>> see your point.  However,
>> there is something very subtle here.  This is dealing with responses
>> and we add the hi-entries once we
>> get the response.  However, the value of the hi-index actually does
>> reflect a chronological order in terms of generating requests.  As
>> discussed previously, you lose that chronological ordering across
>> entities, but for a specific
>> SIP entity they do reflect the chronological processing at that
>> entity.  So, I don't think this is broken.  With the cache
>> model, we aren't adding them to the cache until we get the response
>> even though they were added to the request (in both chronological and
>> lexicographical order).
>> [/MB]
>>
>>
>>     9.3.  Receiving a Response with History-Info or Request Timeouts
>>
>>     Step 3:  Add additional hi-entries
>>
>>        The SIP entity MUST also add to the cache any hi-entries received
>>        in the response that are not already in the cache.  This situation
>>        can occur when the entity that generated the non-100 response
>>        retargeted the request before generating the response.  As per
>>        Step 1, the hi-entries MUST be added to the cache in ascending
>>        order as indicated by the values in the hi-index parameters of the
>>        hi-entries
>>
>> This paragraph needs more extensive changes:
>> [MB] It's not clear to me why you think it needs to be done this way.
>> This is still mapping back to the original request sent by a specific
>> entity (e.g., Proxy A).   I think you are confused by the scenario
>> being described.  The retargeting was not done by Proxy A that is
>> caching the hi-entries.  It was done by Proxy B that received the
>> request and it retargeted before returning the 1xx response thus there
>> is an additional hi-entry that Proxy A should add to its cache. [/MB]
>>
>>     Step 3:  Add and update additional hi-entries
>>
>>        The SIP entity MUST also add to the cache any hi-entries received
>>        in the response that are not already in the cache.  This
>>        can occur when the entity that generated the non-100 response
>>        retargeted the request before generating the response.  Such
>>        hi-entries MUST be added to the end of the cache in the order
>>        they appear in the History-Info header field of the response.
>>
>>        If an hi-entry in the response contains Reason header value(s)
>>        and the matching hi-entry in the cache does not, the Reason
>>        header value(s) MUST be added to the cached hi-entry.  This can
>>        occur when the hi-entry was added to the cache by a provisional
>>        response to an INVITE that was further retargeted, and later a
>>        final response to that INVITE is received.
>>
>>        In these cases, an hi-entry in a response's History-Info header
>>        field and an hi-entry in the cache are considered matching when
>>        they have the same index value and the hi-targeted-to-uris are
>>        equivalent (per RFC 3261 section 19.1.4, but ignoring the Reason
>>        header fields).
>>
>>     10.2.  Reason in the History-Info Header Field
>>
>>     A Reason header field is added to the "headers" component in an hi-
>>     targeted-to-uri when the hi-entry is added to the cache based upon
>> ----------------------------------------^^^^^^^^
>> change to "added to, or updated within,"
>>
>>     the receipt of a non-100 or non-2xx SIP response, as described in
>>     Section 9.3.  [...]
>> [MB] Why "updated within"?  [/MB]
>>
>>     10.3.  Indexing in the History-Info Header Field
>>
>>     In order to maintain ordering and accurately reflect the retargeting
>>     of the request, the SIP entity MUST add a hi-index to each hi-entry.
>>     Per the syntax in Section 5, the hi-index consists of a series of
>>     nonnegative integer separated by dots (e.g., 1.1.2).  Each dot
>>     reflects a SIP forwarding hop.  The nonnegative integer following
>>     each dot reflects the order in which a request was retargeted at the
>>     hop.  The highest nonnegative integer at each hop reflects the number
>>     of entities to which the request has been retargeted at the specific
>>     hop (i.e., the number of branches) at the time that the request
>>     represented by this hi-entry was generated.  Thus, the indexing
>>     results in a logical tree representation for the history of the
>>     request and the hi-entries are given in the preorder of the tree.
>> -----------^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
>> delete this clause
>> [MB] Okay. [/MB]
>>
>> ==========
>>
>>> 3. Due to the confusion discussed in #2, we have to consider whether
>>> consensus was actually found in the WGLC.
>>
>> [MB] We had consensus in WGLC.  Your comments came well after WG last call
>> and after IETF LC.
>>
>>> We thought we had
>>> consensus, but people did not have the same understanding of this
>>> aspect of the draft.  (Ugh.)
>>
>>
>> This is a procedural issue, not a technical one.  I don't see any
>> objection to correcting the normative text if everyone who cares
>> agrees that the changes express what we intended.  (We should check
>> this point with the AD.)
>>
>> [MB] I do not see that we are changing any of the intended normative
>> behavior but rather fixing things that didn't quite reflect the
>> normative behavior as stated elsewhere (i.e., inconsistencies). [/MB]
>>
>> ==========
>>
>>> 5. In draft-ietf-sipcore-rfc4244bis-callflows-01, examples 3.6 and 3.7
>>> do not show the History-Info value in the final 200 response to the
>>> UAC.  This final value would be useful, in particular, because they
>>> give further examples of how proxies process History-Info.
>>
>>
>> I was mistaken here:  The final 200 response from the example.com
>> proxy to the UAC in these examples has the same History-Info as the
>> 200 response arriving at the example.com proxy, because there is no
>> History-Info information that the proxy knows that is not known to the
>> UAS that generated the 200 response.  Thus there is nothing
>> interesting to document.
>>
>> ==========
>>
>>> 6. Interestingly, despite the fact that there can be duplicate index
>>> values, when a request is received by a UAS, there is no ambiguity in
>>> the history when tracing *back* from the last hi-entry (the one that
>>> denotes the UAS).  This is because even if this UAS is "down one fork"
>>> from a forking non-supporting proxy, none of the hi-entries generated
>>> "down the other fork" will be present in the request the UAS sees.
>>>
>>> This is in contrast to the situation at the UAC, which may see
>>> hi-entries with duplicate indexes, and the interpretation of "rc",
>>> "mp", and "np" parameters can be ambiguous.
>>
>>
>> This is an interesting observation, but it does not require
>> documentation in the RFCs.
>>
>> ==========
>>
>>> 7. In example 3.6 message F6 in
>>> draft-ietf-sipcore-rfc4244bis-callflows-01, the History-Info is:
>>>
>>>     History-Info: <sip:bob@example.com>;index=1
>>>     History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302>;\
>>>                        index=1.1;rc=1
>>>     History-Info: <sip:carol@example.com>;index=1.2;mp=1
>>>     History-Info: <sip:carol@192.0.2.4?Reason=SIP%3Bcause%3D408>;\
>>>                        index=1.2.1;rc=1.2
>>>     History-Info: <sip:vm@example.com;\
>>>                        target=sip:bob%40example.com;cause=408>;\
>>>                        index=1.3;mp=1.2
>>>     History-Info: <sip:vm@192.0.2.6;\
>>>                        target=sip:bob%40example.com;cause=408>;\
>>>                        index=1.3.1;rc=1.3
>>>
>>> This is what we intend, as it shows both where the index=1.3 URI came
>>> from (sip:bob@example.com) as well as why the index=1.3 fork was
>>> generated (because sip:carol@example.com failed).  But it points out
>>> that we need to improve the description in rfc4244bis section 5 of
>>> what the values of the "rc" (etc.) parameters are.  A naive reader
>>> might think "rc=1.2" in the 1.3 entry means that
>>> <sip:vm@example.com;target=sip:bob%40example.com;cause=408> was
>>> generated as a contact for <sip:carol@192.0.2.4>.  Instead, it means
>>> that the 1.3 fork was generated *because* the 1.2 fork returned a
>>> response; it was generated *from* the index=1 URI.
>>>
>>> Actually, the real rules for interpreting "rc" (etc.) are not simple:
>>
>> [MB] The above is an example, thus the description is not complete. [/MB]
>>>
>>>
>>> 1) If the index-val in the param points to an entry with a Reason
>>> header-param whose value is "SIP;cause=3xx" (after removing
>>> escaping!), then:
>>> - the URI was one of the Contact values in the 3xx response
>>> - that response was why the new fork was generated.
>>>
>>> 2) If the index-val in the param points to the parent of this
>>> hi-entry, then:
>>> - the URI was generated from the URI of the parent hi-entry
>>> - the fork was not generated due to the response from any other fork.
>>>
>>> 3) If the index-val in the param points to an hi-entry that is not the
>>> parent, and that hi-entry does not contain a Reason header-param
>>> containing a 3xx status, then:
>>> - the URI was generated from the URI of the parent hi-entry
>>> - the fork was generated due to the response from the hi-entry with
>>> the specified index-val.
>>>
>>> And this doesn't match the section 5 definition of the value of "rc"
>>> (etc.), which is:
>>>
>>>           The "rc" header field parameter contains the value of the
>>>           hi-index in the hi-entry with an hi-targeted-to-uri that
>>>           reflects the Request-URI that was retargeted
>>
>>
>> In section 5, these paragraphs need to be revised:
>>
>>     o  hi-target-param: An optional parameter reflecting the mechanism by
>>        which the Request URI captured in the hi-targeted-to-uri in the
>>        History-Info header field value (hi-entry) was determined.  This
>>        parameter is either an "rc", "mp" or "np" header field parameter,
>>        which is interpreted as follows:
>>
>>           "rc": The hi-targeted-to-URI represents a change in Request-URI
>>           while the target user remains the same.  This occurs for
>>           example when user has multiple AoRs as an alias.  The "rc"
>>           header field parameter contains the value of the hi-index in
>>           the hi-entry with an hi-targeted-to-uri that reflects the
>>           Request-URI that was retargeted
>>
>>           "mp": The hi-targeted-to-URI represents a user other than the
>>           target user associated with the Request-URI in the incoming
>>           request that was retargeted.  This occurs when a request is
>>           statically or dynamically retargeted to another user
>>           represented by an AoR unassociated with the AoR of the original
>>           target user.  The "mp" header field parameter contains the
>>           value of the hi-index in the hi-entry with an hi-targeted-to-
>>           uri that reflects the Request-URI that was retargeted, thus
>>           identifying the "mapped from" target.
>>
>>           "np": The hi-targeted-to-URI represents that there was no
>>           change in Request-URI.  This would apply for example when a
>>           proxy merely forwards a request to a next hop proxy and loose
>>           routing is used.  The "np" header field parameter contains the
>>           value of the hi-index in the hi-entry with an hi-targeted-to-
>>           uri that reflects the Request-URI that was copied unchanged
>>           into the request represented by this hi-entry.  That value will
>>           usually be the hi-index of the parent hi-entry of this hi-
>>           entry.
>>
>> The revised paragraphs are:
>>
>> [MB] I don't agree with this proposal - that's way too much detail for
>> this section.  I believe that the normative
>> sections describe how/when you add the "rc" tag which should match
>> what you are describing in your new bullets.    I will note that this
>> section has been updated several times both adding more text and
>> reducing the text in this section.  What you have proposed for the
>> bullets below (i.e., the "rc", "mp" etc.) is some of the level of
>> detail we had previously (in the -02).   I believe the text you
>> suggest is application logic - note that items 2 & 4 describe the "rc"
>> scenarios and this wasn't intended to be comprehensive.  If you can
>> highlight what's missing in those 2 bullets, we can perhaps make some
>> changes there.   I will note if these suggestions had been made much
>> earlier in the process, it would have been much more helpful - much of
>> the text you mention has been in the document since around the -02.
>> We did several rounds of major edits while developing this document
>> both while it was an individual and as a WG document.  Note that this
>> document became a WG document literally 3 years ago.  It's quite
>> disruptive to be raising this issues at this stage. [/MB]
>>
>>     [This first paragraph is unchanged:]
>>     o  hi-target-param: An optional parameter reflecting the mechanism by
>>        which the Request URI captured in the hi-targeted-to-uri in the
>>        History-Info header field value (hi-entry) was determined.  This
>>        parameter is either an "rc", "mp" or "np" header field parameter,
>>        which is interpreted as follows:
>>
>>           "rc": The hi-targeted-to-URI represents a change in Request-URI
>>           while the target user remains the same.  This occurs for
>>           example when user has multiple AoRs as an alias or when
>>           forwarding the request to a registered contact of an AoR.
>>
>>           "mp": The hi-targeted-to-URI represents a user other than the
>>           target user associated with the Request-URI in the incoming
>>           request that was retargeted.  This occurs when a request is
>>           statically or dynamically retargeted to another user
>>           represented by an AoR unassociated with the AoR of the original
>>           target user.
>>
>>           "np": The hi-targeted-to-URI represents that there was no
>>           change in Request-URI.  This would apply for example when a
>>           proxy merely forwards a request to a next hop proxy and loose
>>           routing is used.  The "np" header field parameter contains the
>>           value of the hi-index in the parent hi-entry of this hi-entry.
>>
>>        If the index-val in the parameter points to the parent hi-entry of
>> this
>>        hi-entry, then:
>>        - the URI was generated from the URI of the parent hi-entry
>>        - the fork was not generated due to the response from any other
>> fork.
>>
>>        If the index-val in the parameter points to an hi-entry with a
>> Reason
>>        header value whose protocol is "SIP" and whose cause begins with
>> "3",
>>        then:
>>        - the URI was one of the Contact values in the 3xx response
>>        - the fork was generated due to the 3xx response from the hi-entry
>> with
>>        the specified index-val.
>>
>>        If the index-val in the param points to an hi-entry that is not the
>>        parent, and that hi-entry does not contain a Reason header value
>>        containing a SIP 3xx cause, then:
>>        - the URI was generated from the URI of the parent hi-entry
>>        - the fork was generated due to the response from the hi-entry with
>>        the specified index-val.
>>
>> ======================================================================
>>
>> I believe that is everything to be done.
>>
>> Dale
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore

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Cc: SIPCORE <sipcore@ietf.org>, "R.Jesske@telekom.de" <R.Jesske@telekom.de>, "Dale R. Worley" <worley@ariadne.com>, Laura Liess <laura.liess.dt@googlemail.com>
Subject: Re: [sipcore] Verify draft-ietf-sipcore-rfc4244bis-callflows-02.txt
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Hi Marianne;

 Thank you for further comments.. My comments inline..

On Feb 14, 2013, at 5:11 AM, <marianne.mohali@orange.com> =
<marianne.mohali@orange.com> wrote:

> Hello,
>=20
> Most of my comments have been perfectly addressed but I have still =
some concerns with the following changes:
>=20
> MM#1 =A73.6 in F3: there is an "mp" tag in a Contact header; I guess =
it's a mistake and it should be a History-Info header (as in F4).

 This is intentional. Per section  4.2.1 of RFC 4244bis regarding =
Redirect Server=20
behavior, redirect server would indicate the method a redirect-to =
contact was=20
determined.=20

>=20
> MM#2 =A73.6 in F6: The History-Info header is
> History-Info: <sip:bob@example.com>;index=3D1 			=
[MM] ok [/MM]
> History-Info: <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302>;\
>      		   index=3D1.1;rc=3D1 				[MM] ok =
[/MM]
> History-Info: <sip:carol@example.com>;index=3D1.2;mp=3D1 	[MM] ok =
[/MM]
> History-Info: <sip:carol@192.0.2.4?Reason=3DSIP%3Bcause%3D480>;\
>                     index=3D1.2.1;rc=3D1.2 				=
[MM] nok, it is a cause URI parameter that should be set to "480" (call =
deflection) following RFC4458. The Reason header is escaped in the SIP =
URI only when it is a real received respond (it is the case with the =
302) [/MM]
> History-Info: <sip:vm@example.com;\
>                     target=3Dsip:bob%40example.com>;\
>                     index=3D1.3;mp=3D1 				=
[MM] nok, The hi-entry with the voicemail should be a copy of the R-URI =
so with the cause=3D480 URI parameter. Additionally, the vm is finally =
reached because Carol does not reply, so the no reply reason (cause=3D408)=
 should also be reflected in a cause URI parameter following RFC4458. =
Hence, the hi-entry should be History-Info: =
<sip:vm@example.com;target=3Dsip:bob%40example.com;cause=3D480;cause=3D408=
>;                    index=3D1.3;mp=3D1  =3D> Oops, I think we have a =
problem... This is due to the fact that at the same time we have =
successive forwarding (1 after the other) with Carol at the =
second-to-last position and we try to join B's vm (1st diverting user). =
IMO, we should add some text stating that depending on which voicemail =
is reached (Bob or Carol), the call forwarding reason (Bob or Carol's =
diverting reason) in the cause URI parameter should be set accordingly =
(to be consistent). Here the target vm is Bob so only cause=3D480 should =
be used and Carol's reason for not answering the call is lost. [/MM]
> [MM]If someone have a better/other idea, please don't hesitate... =
[/MM]
> History-Info: <sip:vm@192.0.2.6;\
>                     target=3Dsip:bob%40example.com>;\
>                     index=3D1.3.1;rc=3D1.3 				=
[MM] nok, same comment
>=20
> But IMO it should be:
>=20
> History-Info: <sip:bob@example.com>;index=3D1=20
> History-Info: <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302>;\=20
>                     index=3D1.1;rc=3D1        =20
> History-Info: <sip:carol@example.com>;index=3D1.2;mp=3D1=20
> History-Info: <sip:carol@192.0.2.4;cause=3D480>;\
>                     index=3D1.2.1;rc=3D1.2=20
> History-Info: <sip:vm@example.com;\
>                     target=3Dsip:bob%40example.com;cause=3D480>;\
>                     index=3D1.3;mp=3D1
> History-Info: <sip:vm@192.0.2.6;\
>                     target=3Dsip:bob%40example.com;cause=3D480>;\
>                     index=3D1.3.1;rc=3D1.3

 Ok. Fixed.

>=20
> MM#3 =A73.7 in F4 and followings:=20
> In the hi-entry the reason-text has been added.=20
> I have 2 comments:=20
> 1. why adding the reason-text? (and why not ;)=3D> ok for me=20
> 2. if the reason-text is added, it should be "escaped" in the same way =
as the cause parameter. =20
> So that, the hi-entry:
> History-Info: <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302\
>                         ;text=3D"Moved Temporarily">\
>                 ;index=3D1.1;rc=3D1
>=20
> Should be:
> History-Info: <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302\
>                         %3Btext%3D"Moved Temporarily">\
>                 ;index=3D1.1;rc=3D1

 Good catch. Thanks

 Shida

>=20
> Here is my feedback.
>=20
> Best Regards,
> Marianne
>=20
> -----Message d'origine-----
> De : sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] De la =
part de Paul Kyzivat
> Envoy=E9 : vendredi 1 f=E9vrier 2013 23:37
> =C0 : R.Jesske@telekom.de; Laura Liess; MOHALI Marianne OLNC/OLN; =
Brett Tate; Dale R. Worley
> Cc : SIPCORE
> Objet : [sipcore] Verify =
draft-ietf-sipcore-rfc4244bis-callflows-02.txt
>=20
> A new version of 4244bis-callflows has been published, to address the =
comments we received in the WGLC review at the end of last year.
>=20
> I've sent this *To* the people who provided comments.
> I would appreciate your verifying that your comments were properly =
addressed. (Please comment back one way or the other.)
>=20
> This is *not* a new WGLC. So please don't bring up *new* issues now. =
We are past that. I just want to verify that the comments already raised =
have been handled. We are holding the bis to publish concurrently with =
the callflows, so is the last thing holding them up.
>=20
> 	Thanks,
> 	Paul
>=20
>=20
> -------- Original Message --------
> Subject: [sipcore] I-D Action:=20
> draft-ietf-sipcore-rfc4244bis-callflows-02.txt
> Date: Tue, 29 Jan 2013 12:49:12 -0800
> From: internet-drafts@ietf.org
> To: i-d-announce@ietf.org
> CC: sipcore@ietf.org
>=20
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
>  This draft is a work item of the Session Initiation Protocol Core =
Working Group of the IETF.
>=20
> 	Title           : Session Initiation Protocol (SIP) History-Info =
Header=20
> Call Flow Examples
> 	Author(s)       : Mary Barnes
>                           Francois Audet
>                           Shida Schubert
>                           Detecon International Gmbh
>                           Christer Holmberg
> 	Filename        : draft-ietf-sipcore-rfc4244bis-callflows-02.txt
> 	Pages           : 46
> 	Date            : 2013-01-29
>=20
> Abstract:
>    This document describes use cases and documents call flows which
>    require the History-Info header field to capture the Request-URIs =
as
>    a Session Initiation Protocol (SIP) Request is retargeted.  The use
>    cases are described along with the corresponding call flow diagrams
>    and messaging details.
>=20
>=20
> The IETF datatracker status page for this draft is:
> =
https://datatracker.ietf.org/doc/draft-ietf-sipcore-rfc4244bis-callflows
>=20
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-sipcore-rfc4244bis-callflows-02
>=20
> A diff from the previous version is available at:
> =
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-sipcore-rfc4244bis-callflows=
-02
>=20
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>=20
>=20
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>=20
> =
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Subject: Re: [sipcore] Verify draft-ietf-sipcore-rfc4244bis-callflows-02.txt
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Hi Brett;

 Thanks for further comments.=20

My comments inline.

On Feb 9, 2013, at 11:50 AM, Brett Tate wrote:

> Howdy,
>=20
> Most of my comments have been adequately addressed.
>=20
> Thanks,
> Brett
>=20
> -----
>=20
> Concerning my various comments to include Max-Forwards, the values are =
good enough for me; however I was surprised to see some of the =
decrementing and initial values.  For instance, section 3.1 F5 and F13 =
contain initial values within ACK reflecting the related INVITE instead =
of 70.  And for instance, section 3.1 F6 and F9 reflect the proxy =
decrementing the value again for INVITE sent to new location because of =
302 or timeout.

My bad. Will fix these.=20

>=20
> If not intentional, the figure labels are of the format "Figure 1: =
Figure 1." and not consistently located (i.e. before or after detailed =
messages).

I will fix these as well.=20

>=20
> Section 3.2 F3: top Via received should be on the second via.

Thanks for the catch. Done.=20

>=20
> Section 3.4 picture: 302 ACK arrow reflection wrong source.  The same =
applies to sections 3.6 and 3.7.

Thanks. Done.

>=20
> Concerning =
http://www.ietf.org/mail-archive/web/sipcore/current/msg05265.html =
comment 2, section 3.5 was not modified; thus I'll assume that the =
transport switch from UDP to TCP was intentional (i.e. registered UDP =
Contact but INVITE sent over TCP).

I must have overlooked this.. Fixed.=20

>=20
> Section 3.9 F2: Added Max-Forwards to response.

Thanks. Fixed.

Regards
 Shida

>=20
> New comment (if want to address): figure 1 indicates a "retransmit =
INVITE" with arrow after having received a 180.=20
>=20
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Hi Dale;

 Thanks for further comments..

On Feb 4, 2013, at 12:53 PM, Dale R. Worley wrote:

> My previous comments have been addressed acceptably.  But there are a
> few new nits that I've spotted:
> 
> ** General
> 
> * On page 19
> 
>   Alice       example.com     Gold          Silver       Agent
> 
>   |              |              |             |            |
>   | INVITE F1    |              |             |            |
>   |------------->|              |             |            |
>   |              |              |             |            |
>   |              |  INVITE F2   |             |            |
>   |              |------------->|             |            |
>   |              |              |             |            |
>   |              |  302 Moved Temporarily F3  |            |
>   |              |<-------------|             |            |
>   |              |              |             |            |
>   |             ACK             |             |            |
>   |---------------------------->|             |            |
>   |              |              |             |            |
> 
> The ACK should be shown thus:
> 
>   |              |              |             |            |
>   |              |  ACK         |             |            |
>   |              |------------->|             |            |
>   |              |              |             |            |
> 
> * On page 27, the top ACK should be shown thus:
> 
>   |              |              |             |            |
>   |              |  ACK         |             |            |
>   |              |------------->|             |            |
>   |              |              |             |            |
> 
> * On page 31, the top ACK should be shown thus:
> 
>   |              |              |             |            |
>   |              |  ACK         |             |            |
>   |              |------------->|             |            |
>   |              |              |             |            |
> 
> * On page 40, there are two occurrences of:
> 
>   To: <sip:sip:tgruu.7hs==jd7vnzga5w7fajsc7-ajd6fabz0f8g5@example.com\
>    ;gr>
> 
> "sip:sip:" should be replaced with "sip:".

DONE.

> 
> ** http://www.ietf.org/mail-archive/web/sipcore/current/msg05489.html
> 
> * "5. In draft-ietf-sipcore-rfc4244bis-callflows-01, examples 3.6 and 3.7
> do not show the History-Info value in the final 200 response to the
> UAC.  This final value would be useful, in particular, because they
> give further examples of how proxies process History-Info."
> 
> In neither of these examples is the final 200 response shown.  I
> assume that this is the authors' intention.

 This, I think was because the value of h-i would not change from 
the last proxy to UAC, and didn't see a need. If you want to see this 
added. I will add them

> 
> ** http://www.ietf.org/mail-archive/web/sipcore/current/msg05429.html
> 
> * "But in draft-ietf-sipcore-rfc4244bis-callflows-01, Reason header
> values for 1xx responses are not generated.  See message F8 in section
> 3.1."
> 
> rfc4244bis-callflows-02 does not show Reason headers with 1xx values.
> I expect this is because the authors intend for this to be how
> History-Info operates.  But that does mean we are committed to
> updating rfc4244bis to state that (because it currently says that 1xx
> responses set Reason headers).

 Okay, I will add these. 

> 
> Dale
> _______________________________________________
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> From: Shida Schubert <shida@ntt-at.com>

[everything you write is fine with me]

> > * "5. In draft-ietf-sipcore-rfc4244bis-callflows-01, examples 3.6 and 3.7
> > do not show the History-Info value in the final 200 response to the
> > UAC.  This final value would be useful, in particular, because they
> > give further examples of how proxies process History-Info."
> > 
> > In neither of these examples is the final 200 response shown.  I
> > assume that this is the authors' intention.
> 
>  This, I think was because the value of h-i would not change from 
> the last proxy to UAC, and didn't see a need. If you want to see this 
> added. I will add them

I agree that there is no need.  I believe I was under a
misapprehension when I wrote point 5.

Dale

From marianne.mohali@orange.com  Wed Feb 20 07:45:05 2013
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From: <marianne.mohali@orange.com>
To: "Dale R. Worley" <worley@ariadne.com>
Thread-Topic: [sipcore] Verify draft-ietf-sipcore-rfc4244bis-callflows-02.txt
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Date: Wed, 20 Feb 2013 15:44:56 +0000
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Subject: Re: [sipcore] Verify draft-ietf-sipcore-rfc4244bis-callflows-02.txt
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Hi Dale,

See my answer inline [MM2]

Marianne

> -----Message d'origine-----
> De=A0: Dale R. Worley [mailto:worley@ariadne.com]
> Envoy=E9=A0: lundi 18 f=E9vrier 2013 22:06
> =C0=A0: MOHALI Marianne OLNC/OLN
> Cc=A0: pkyzivat@alum.mit.edu; R.Jesske@telekom.de;
> laura.liess.dt@googlemail.com; brett@broadsoft.com; sipcore@ietf.org
> Objet=A0: Re: [sipcore] Verify draft-ietf-sipcore-rfc4244bis-callflows-
> 02.txt
>=20
> > From: <marianne.mohali@orange.com>
> > Subject: RE: [sipcore] Verify
> > draft-ietf-sipcore-rfc4244bis-callflows-02.txt
> >
> > MM#1 B'3.6 in F3: there is an "mp" tag in a Contact header; I guess
> > it's a mistake and it should be a History-Info header (as in F4).
>=20
> No, that's to show that the contact <sip:carol@example.com> was
> generated by "mapping" from the URI with index 1, <sip:bob@example.com>.
> This is described in 4244bis section 8.
> (There are a number of such Contact headers in the callflows.)
>=20

[MM2] Right, my mistake




> > MM#2 B'3.6 in F6: The History-Info header is
> > History-Info: <sip:bob@example.com>;index=3D1 			[MM] ok [/MM]
> > History-Info: <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302>;\
> > 		   index=3D1.1;rc=3D1 				[MM] ok [/MM]
> > History-Info: <sip:carol@example.com>;index=3D1.2;mp=3D1 	[MM] ok [/MM]
> > History-Info: <sip:carol@192.0.2.4?Reason=3DSIP%3Bcause%3D480>;\
> > 		     index=3D1.2.1;rc=3D1.2
> > [MM] nok, it is a cause URI parameter that should be set to "480"
> > (call deflection) following RFC4458. The Reason header is escaped in
> > the SIP URI only when it is a real received respond (it is the case
> > with the 302) [/MM]
>=20
> I believe the cause is supposed to be 408 (Timeout) rather than 480,
> and it is correct for the proxy to synthesize a 408 cause here because
> of 4244bis section 10.2 paragraph 2.  (And many SIP systems treat
> timing out while waiting for a response as if they received an explicit
> 408 response.)
>=20
> This mistake of 480 for 408 is also present in the request-line of F6
> and the History-Info of F7.
>=20
> (Confusion can easily arise because 480 (Temporarily Unavailable) is
> also a plausible cause for diversion to VM.)
>=20

[MM2] No, there is no mistake in the code, the correction were done after o=
ne of my comments. Here we are in a Call Deflection immediate response situ=
ation (as defined in TS24.604) and the cause URI parameter value to be used=
 and defined in RFC4458 is 480. My point was about the format which is a [R=
eason header's cause parameter] format instead of the [cause URI parameter]=
 format in the URI.

The line should be:
History-Info: <sip:carol@192.0.2.4;cause=3D480>;\
 		 index=3D1.2.1;rc=3D1.2=20

>From RFC4458
                +---------------------------------+-------+
                | Redirecting Reason              | Value |
                +---------------------------------+-------+
                | Unknown/Not available           | 404   |
                | User busy                       | 486   |
                | No reply                        | 408   |
                | Unconditional                   | 302   |
                | Deflection during alerting      | 487   |
                | Deflection immediate response   | 480   |
                | Mobile subscriber not reachable | 503   |
                +---------------------------------+-------+




> > History-Info: <sip:vm@example.com;\
> > 		     target=3Dsip:bob%40example.com>;\
> > 		     index=3D1.3;mp=3D1
> > [MM] nok, The hi-entry with the voicemail should be a copy of the
> > R-URI so with the cause=3D480 URI parameter. Additionally, the vm is
> > finally reached because Carol does not reply, so the no reply reason
> > (cause=3D408) should also be reflected in a cause URI parameter
> > following RFC4458. Hence, the hi-entry should be History-Info:
> > <sip:vm@example.com;target=3Dsip:bob%40example.com;cause=3D480;cause=3D=
408>;
> > index=3D1.3;mp=3D1  =3D> Oops, I think we have a problem... This is due=
 to
> > the fact that at the same time we have successive forwarding (1 after
> > the other) with Carol at the second-to-last position and we try to
> > join B's vm (1st diverting user). IMO, we should add some text
> stating
> > that depending on which voicemail is reached (Bob or Carol), the call
> > forwarding reason (Bob or Carol's diverting reason) in the cause URI
> > parameter should be set accordingly (to be consistent). Here the
> > target vm is Bob so only cause=3D480 should be used and Carol's reason
> > for not answering the call is lost. [/MM] [MM]If someone have a
> > better/other idea, please don't hesitate... [/MM]
>=20
> Beware that message F6 does not correspond to entry 1.3 but rather to
> entry 1.3.1, which is derived from 1.3 by internal redirection.  (It
> might be worth adding a note in the text to warn the reader!)  So it is
> entry 1.3.1 that must have the cause parameter, to match the request-
> URI.
>=20
> However, since the 1.3.1 request-URI has a cause parameter, it is
> likely that 1.3 request-URI has the same cause parameter.
>=20
> The messier problem is not really present.  There is only one cause at
> this point, the implicit 408 response from Carol.  And there may be
> complexities concerning what the diversion URI should be, but that is
> the proxy's problem, not a problem in designing 4244bis, and we do not
> have to solve it here, we only have to show the History-Info that comes
> from this particular proxy's decisions.
>=20

[MM2] OK, so the following hi-entry (index=3D1.3.1) must contain a cause UR=
I parameter set to "cause=3D408" (removed from the draft-01 version)


> > History-Info: <sip:vm@192.0.2.6;\
> > 		     target=3Dsip:bob%40example.com>;\
> > 		     index=3D1.3.1;rc=3D1.3
> > [MM] nok, same comment
> >
> > But IMO it should be:
> >
> > History-Info: <sip:bob@example.com>;index=3D1
> > History-Info: <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302>;\
> > 		     index=3D1.1;rc=3D1
> > History-Info: <sip:carol@example.com>;index=3D1.2;mp=3D1
> > History-Info: <sip:carol@192.0.2.4;cause=3D480>;\
> > 		     index=3D1.2.1;rc=3D1.2
> > History-Info: <sip:vm@example.com;\
> > 		     target=3Dsip:bob%40example.com;cause=3D480>;\
> > 		     index=3D1.3;mp=3D1
> > History-Info: <sip:vm@192.0.2.6;\
> > 		     target=3Dsip:bob%40example.com;cause=3D480>;\
> > 		     index=3D1.3.1;rc=3D1.3
> >



> > MM#3 B'3.7 in F4 and followings:
> > In the hi-entry the reason-text has been added.
> > I have 2 comments:
> > 1. why adding the reason-text? (and why not ;)=3D> ok for me 2. if the
> > reason-text is added, it should be "escaped" in the same way as the
> > cause parameter.
> > So that, the hi-entry:
> > History-Info: <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302\
> > 			 ;text=3D"Moved Temporarily">\
> > 		 ;index=3D1.1;rc=3D1
> >
> > Should be:
> > History-Info: <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302\
> > 			 %3Btext%3D"Moved Temporarily">\
> > 		 ;index=3D1.1;rc=3D1
>=20
> The header field as it stands is:
>=20
>    History-Info: <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302\
>                               ;text=3D"Moved Temporarily">\
>                  ;index=3D1.1;rc=3D1
>=20
> but the escaping is incorrect.  The Reason header would be:
>=20
>     Reason: SIP;cause=3D302;text=3D"Moved Temporarily"
>=20
> so the 'hname' is 'Reason' and the 'hvalue' is
> 'SIP;cause=3D302;text=3D"Moved Temporarily"', so the URI should be
>=20
> <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302%3Btext%3D%34Moved%20Tempora
> rily%34>
>=20
> There are 4 instances of this problem, in 3.7 F4, F5, F6, and F7
>=20

[MM2] Exact!

> Dale

[MM2]Marianne

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From worley@shell01.TheWorld.com  Wed Feb 20 07:45:12 2013
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From: worley@ariadne.com (Dale R. Worley)
Sender: worley@ariadne.com (Dale R. Worley)
To: Mary Barnes <mary.ietf.barnes@gmail.com>
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Cc: sipcore@ietf.org
Subject: Re: [sipcore] Fwd: Updates to draft-ietf-sipcore-rfc4244bis-11, version 1.
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> From: Mary Barnes <mary.ietf.barnes@gmail.com>
> 
> On Tue, Feb 19, 2013 at 10:50 AM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
> >
> > This is very hard to follow.
> > But I do get it that the two of you are getting closer but still not in
> > total agreement. Can the two of you reach a mutually satisfactory
> > conclusion?

Sorry about how messy the discussion has gotten.  But the two of us do
understand what the proposed changes are about and we're trying to get
them settled as quickly as possible.

> > I'd love to have a new version with a direct diff to evaluate it all.
> >
> > I must say that I am surprised that so much effort was put into the
> > hierarchical numbering scheme for entries and yet the conclusion is that the
> > entries are not sorted in accord with this hierarchy. (Or did I get that
> > wrong?)
> 
> [MB] As I highlighted to Dale, within an entity they are in order.
> The issue is due to the fact that not all entities will support HI,
> thus you cannot guarantee that the lexicographical order is meaningful
> (amongst all the entities).  Since we do require an entity to add the
> entries in chronological order using the indexing scheme as specifed
> they are in order for that entity.  When the request is forwarded and
> a response contains additional hi-entires (due to retargeting at the
> next hop), the ordering is guaranteed for the entity to which the
> request was forwarded.  The issue comes when you get a reset to 0
> indicating an entity did not support HI, then you lose any ordering.
> [/MB]

There are some unusual, tricky cases where the hi-entries in an
entity's cache won't be in lexicographic order.  They involve parallel
forking, where the processing on each fork sends back multiple 1xx
provisional responses carrying additional retargetings.

But in ordinary cases, adding to the cache
sequentially/chronologically is *much* easier to specify than
specifying the processing that would keep lexicographic ordering in
all cases.  As Mary points out, the problem is with forking by
non-supporting intermediate entities, which results in duplicate index
values (containing "0"), so you can't just sort by the index values.
(There additional hacks that could get around this problem in most
cases, but the additional complexity suggests that the chronological
approach is simpler.)

The penalty is that in parallel forking cases where the forking proxy
is non-supporting, the application logic at the *UAC* end may not be
able to reliably reconstruct the forking structure, due to the
duplicate index values and the interleaving of responses from the
parallel forks.

However (I mention this in a note buried in my earlier message), at
the *UAS* end, this messiness isn't seen:  A UAS won't see hi-entries
that come from sister forks originating at a non-supporting proxy, and
so its interpretation of the History-Info won't be corrupted.  And it
is the interpretation by the UAS that we really care about.

Dale

From oferg@avaya.com  Wed Feb 20 09:54:41 2013
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From: Ofer Goren <oferg@avaya.com>
To: "Roy, Radhika R CIV USARMY (US)" <radhika.r.roy.civ@mail.mil>
Thread-Topic: question about B2B and Forking (UNCLASSIFIED)
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Subject: Re: [sipcore] question about B2B and Forking (UNCLASSIFIED)
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Hi Roy.
This is exactly my question: is it needed. It looks like it might be, but I=
 cannot understand why would anybody wants to do that. It either a B2BUA, i=
n that case it holds a state machine for each call created on each side, OR=
 it is a proxy forwarding transactions from side to side. I don't understan=
d why anyone would like to mix the behavior, but I can't find anything in t=
he specs prohibit it.

Ofer.

-----Original Message-----
From: Roy, Radhika R CIV USARMY (US) [mailto:radhika.r.roy.civ@mail.mil]=20
Sent: Tuesday, February 19, 2013 5:18 PM
To: Ofer Goren
Cc: sipcore@ietf.org
Subject: RE: question about B2B and Forking (UNCLASSIFIED)

Classification: UNCLASSIFIED
Caveats: NONE

Ofer:

In this case it not B2BUA spec per se. It will be rather specific to
"Forking Specifications using B2BUA." If we want to have a interoperable
implementation of forking using B2BUA. Is it really needed?

BR/Radhika


-----Original Message-----
From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf
Of Ofer Goren
Sent: Tuesday, February 19, 2013 8:02 AM
To: sipcore@ietf.org
Subject: [sipcore] question about B2B and Forking

Hi All.

I know B2B behavior is not defined with too much details in the relevant SI=
P
RFCs. I do ask for your opinion:

In forking scenarios, if B2B forks an INVITE to several UAS-es, should it
forward downstream all the provisional responses to the generating UAC? Or,
should it forward only one," absorbing" the rest? Should it even wait for
provisional responses, or should it generates its own toward the UAC,
independent to the upstream side of the session?

If it should forward all provisional responses downstream, it in fact
creates several early dialogs for which it should also handle transactions
such as PRACK (in case of reliable provisional responses), or UPDATEs and
such.=20

=20

So what do you think a B2B should do?

=20

Thanks!

Ofer

=20

=20

_______________________________________________________________

If you lend someone $20, and never see that person again, it was probably
worth the investment.=20

=20

=20


Classification: UNCLASSIFIED
Caveats: NONE



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From: Ofer Goren <oferg@avaya.com>
To: Brett Tate <brett@broadsoft.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: question about B2B and Forking
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Hi Brett.

Why do you think those 3 RFCs suggest that B2BUAs should act as proxies for=
 UPDATE and 18x reliable responses? I'm not sure I follow.

Ofer

From: Brett Tate [mailto:brett@broadsoft.com]
Sent: Sunday, February 17, 2013 7:30 PM
To: Ofer Goren; sipcore@ietf.org
Subject: RE: question about B2B and Forking

Hi Ofer,

The appropriate behavior depends upon the functionality that the B2BUA is p=
roviding.

Within normal call originations with devices compliant to RFC 3261, RFC 326=
2, and RFC 3311, it is typically best to act more like a proxy.

However since there are still devices that don't support RFC 3262, RFC 3311=
, or forking proxies, the B2BUA might have to act differently for interoper=
ability reasons.  Similarly, the B2BUA may need to act differently for serv=
ice reasons.


From: sipcore-bounces@ietf.org<mailto:sipcore-bounces@ietf.org> [mailto:sip=
core-bounces@ietf.org] On Behalf Of Ofer Goren
Sent: Sunday, February 17, 2013 11:23 AM
To: sipcore@ietf.org<mailto:sipcore@ietf.org>
Subject: [sipcore] question about B2B and Forking

Hi All.
I know B2B behavior is not defined with too much details in the relevant SI=
P RFCs. I do ask for your opinion:
In forking scenarios, if B2B forks an INVITE to several UAS-es, should it f=
orward downstream all the provisional responses to the generating UAC? Or, =
should it forward only one," absorbing" the rest? Should it even wait for p=
rovisional responses, or should it generates its own toward the UAC, indepe=
ndent to the upstream side of the session?
If it should forward all provisional responses downstream, it in fact creat=
es several early dialogs for which it should also handle transactions such =
as PRACK (in case of reliable provisional responses), or UPDATEs and such.

So what do you think a B2B should do?

Thanks!
Ofer


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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Hi Brett.<o:p></o:p></=
span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Why do you think those=
 3 RFCs suggest that B2BUAs should act as proxies for UPDATE and 18x reliab=
le responses? I&#8217;m not sure I follow.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Ofer <o:p></o:p></span=
></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Brett Ta=
te [mailto:brett@broadsoft.com]
<br>
<b>Sent:</b> Sunday, February 17, 2013 7:30 PM<br>
<b>To:</b> Ofer Goren; sipcore@ietf.org<br>
<b>Subject:</b> RE: question about B2B and Forking<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Hi Ofer,<o:p></o:p></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">The appropriate behavi=
or depends upon the functionality that the B2BUA is providing.<o:p></o:p></=
span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Within normal call ori=
ginations with devices compliant to RFC 3261, RFC 3262, and RFC 3311, it is=
 typically best to act more like a proxy.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">However since there ar=
e still devices that don&#8217;t support RFC 3262, RFC 3311, or forking pro=
xies, the B2BUA might have to act differently for interoperability reasons.=
&nbsp; Similarly, the B2BUA may need to act differently
 for service reasons.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">
<a href=3D"mailto:sipcore-bounces@ietf.org">sipcore-bounces@ietf.org</a> [<=
a href=3D"mailto:sipcore-bounces@ietf.org">mailto:sipcore-bounces@ietf.org<=
/a>]
<b>On Behalf Of </b>Ofer Goren<br>
<b>Sent:</b> Sunday, February 17, 2013 11:23 AM<br>
<b>To:</b> <a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
<b>Subject:</b> [sipcore] question about B2B and Forking<o:p></o:p></span><=
/p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Hi All.<o:p></o:p></p>
<p class=3D"MsoNormal">I know B2B behavior is not defined with too much det=
ails in the relevant SIP RFCs. I do ask for your opinion:<o:p></o:p></p>
<p class=3D"MsoNormal">In forking scenarios, if B2B forks an INVITE to seve=
ral UAS-es, should it forward downstream all the provisional responses to t=
he generating UAC? Or, should it forward only one,&#8221; absorbing&#8221; =
the rest? Should it even wait for provisional
 responses, or should it generates its own toward the UAC, independent to t=
he upstream side of the session?<o:p></o:p></p>
<p class=3D"MsoNormal">If it should forward all provisional responses downs=
tream, it in fact creates several early dialogs for which it should also ha=
ndle transactions such as PRACK (in case of reliable provisional responses)=
, or UPDATEs and such.
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">So what do you think a B2B should do?<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Thanks!<o:p></o:p></p>
<p class=3D"MsoNormal">Ofer<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</body>
</html>

--_000_D9AD3D1627F52341B829F32E6DDE089149E0CAF6rvilmail2010_--

From brett@broadsoft.com  Wed Feb 20 10:31:31 2013
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From: Brett Tate <brett@broadsoft.com>
To: Ofer Goren <oferg@avaya.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: question about B2B and Forking
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I don't.  I indicated that it is typically best for B2BUA to act more like =
a proxy (i.e. to avoid breaking things that work with proxies) unless there=
 are service or interoperability reasons to do otherwise.

The following STRAW draft may be interest to you since there are all kinds =
of B2BUAs: draft-ietf-straw-b2bua-taxonomy.



From: Ofer Goren [mailto:oferg@avaya.com]
Sent: Wednesday, February 20, 2013 12:59 PM
To: Brett Tate; sipcore@ietf.org
Subject: RE: question about B2B and Forking

Hi Brett.

Why do you think those 3 RFCs suggest that B2BUAs should act as proxies for=
 UPDATE and 18x reliable responses? I'm not sure I follow.

Ofer

From: Brett Tate [mailto:brett@broadsoft.com]
Sent: Sunday, February 17, 2013 7:30 PM
To: Ofer Goren; sipcore@ietf.org<mailto:sipcore@ietf.org>
Subject: RE: question about B2B and Forking

Hi Ofer,

The appropriate behavior depends upon the functionality that the B2BUA is p=
roviding.

Within normal call originations with devices compliant to RFC 3261, RFC 326=
2, and RFC 3311, it is typically best to act more like a proxy.

However since there are still devices that don't support RFC 3262, RFC 3311=
, or forking proxies, the B2BUA might have to act differently for interoper=
ability reasons.  Similarly, the B2BUA may need to act differently for serv=
ice reasons.


From: sipcore-bounces@ietf.org<mailto:sipcore-bounces@ietf.org> [mailto:sip=
core-bounces@ietf.org] On Behalf Of Ofer Goren
Sent: Sunday, February 17, 2013 11:23 AM
To: sipcore@ietf.org<mailto:sipcore@ietf.org>
Subject: [sipcore] question about B2B and Forking

Hi All.
I know B2B behavior is not defined with too much details in the relevant SI=
P RFCs. I do ask for your opinion:
In forking scenarios, if B2B forks an INVITE to several UAS-es, should it f=
orward downstream all the provisional responses to the generating UAC? Or, =
should it forward only one," absorbing" the rest? Should it even wait for p=
rovisional responses, or should it generates its own toward the UAC, indepe=
ndent to the upstream side of the session?
If it should forward all provisional responses downstream, it in fact creat=
es several early dialogs for which it should also handle transactions such =
as PRACK (in case of reliable provisional responses), or UPDATEs and such.

So what do you think a B2B should do?

Thanks!
Ofer


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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">I don&#8217;t.&nbsp; I=
 indicated that it is typically best for B2BUA to act more like a proxy (i.=
e. to avoid breaking things that work with proxies) unless there are servic=
e or interoperability reasons to do otherwise.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">The following STRAW dr=
aft may be interest to you since there are all kinds of B2BUAs: draft-ietf-=
straw-b2bua-taxonomy.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Ofer Gor=
en [mailto:oferg@avaya.com]
<br>
<b>Sent:</b> Wednesday, February 20, 2013 12:59 PM<br>
<b>To:</b> Brett Tate; sipcore@ietf.org<br>
<b>Subject:</b> RE: question about B2B and Forking<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Hi Brett.<o:p></o:p></=
span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Why do you think those=
 3 RFCs suggest that B2BUAs should act as proxies for UPDATE and 18x reliab=
le responses? I&#8217;m not sure I follow.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Ofer <o:p></o:p></span=
></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> Brett Ta=
te [<a href=3D"mailto:brett@broadsoft.com">mailto:brett@broadsoft.com</a>]
<br>
<b>Sent:</b> Sunday, February 17, 2013 7:30 PM<br>
<b>To:</b> Ofer Goren; <a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org=
</a><br>
<b>Subject:</b> RE: question about B2B and Forking<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Hi Ofer,<o:p></o:p></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">The appropriate behavi=
or depends upon the functionality that the B2BUA is providing.<o:p></o:p></=
span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Within normal call ori=
ginations with devices compliant to RFC 3261, RFC 3262, and RFC 3311, it is=
 typically best to act more like a proxy.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">However since there ar=
e still devices that don&#8217;t support RFC 3262, RFC 3311, or forking pro=
xies, the B2BUA might have to act differently for interoperability reasons.=
&nbsp; Similarly, the B2BUA may need to act differently
 for service reasons.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">
<a href=3D"mailto:sipcore-bounces@ietf.org">sipcore-bounces@ietf.org</a> [<=
a href=3D"mailto:sipcore-bounces@ietf.org">mailto:sipcore-bounces@ietf.org<=
/a>]
<b>On Behalf Of </b>Ofer Goren<br>
<b>Sent:</b> Sunday, February 17, 2013 11:23 AM<br>
<b>To:</b> <a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
<b>Subject:</b> [sipcore] question about B2B and Forking<o:p></o:p></span><=
/p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Hi All.<o:p></o:p></p>
<p class=3D"MsoNormal">I know B2B behavior is not defined with too much det=
ails in the relevant SIP RFCs. I do ask for your opinion:<o:p></o:p></p>
<p class=3D"MsoNormal">In forking scenarios, if B2B forks an INVITE to seve=
ral UAS-es, should it forward downstream all the provisional responses to t=
he generating UAC? Or, should it forward only one,&#8221; absorbing&#8221; =
the rest? Should it even wait for provisional
 responses, or should it generates its own toward the UAC, independent to t=
he upstream side of the session?<o:p></o:p></p>
<p class=3D"MsoNormal">If it should forward all provisional responses downs=
tream, it in fact creates several early dialogs for which it should also ha=
ndle transactions such as PRACK (in case of reliable provisional responses)=
, or UPDATEs and such.
<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">So what do you think a B2B should do?<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Thanks!<o:p></o:p></p>
<p class=3D"MsoNormal">Ofer<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</div>
</body>
</html>

--_000_576A8B541C219D4E9CEB1DF8C19C7B881969A535MBX06citservers_--

From worley@shell01.TheWorld.com  Wed Feb 20 12:50:03 2013
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From: worley@ariadne.com (Dale R. Worley)
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To: <marianne.mohali@orange.com>
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Cc: sipcore@ietf.org, R.Jesske@telekom.de, laura.liess.dt@googlemail.com
Subject: Re: [sipcore] Verify draft-ietf-sipcore-rfc4244bis-callflows-02.txt
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The History-Info shown in example 3.6 F6 (of the -02 version) is:

  History-Info: <sip:bob@example.com>;index=1
  History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302>;\
                     index=1.1;rc=1
  History-Info: <sip:carol@example.com>;index=1.2;mp=1
  History-Info: <sip:carol@192.0.2.4?Reason=SIP%3Bcause%3D480>;\
                     index=1.2.1;rc=1.2
  History-Info: <sip:vm@example.com;\
                     target=sip:bob%40example.com>;\
                     index=1.3;mp=1
  History-Info: <sip:vm@192.0.2.6;\
                     target=sip:bob%40example.com>;\
                     index=1.3.1;rc=1.3

In regard to this example Marianne Mohali has written (and I have
heavily abridged):

> From: <marianne.mohali@orange.com>

> [MM2] No, there is no mistake in the code, the correction were done
> after one of my comments. Here we are in a Call Deflection immediate
> response situation (as defined in TS24.604) and the cause URI
> parameter value to be used and defined in RFC4458 is 480. My point
> was about the format which is a [Reason header's cause parameter]
> format instead of the [cause URI parameter] format in the URI.
> 
> The line should be:
> History-Info: <sip:carol@192.0.2.4;cause=480>;\
>  		 index=1.2.1;rc=1.2 
> 
> >From RFC4458
>                 +---------------------------------+-------+
>                 | Redirecting Reason              | Value |
>                 +---------------------------------+-------+
>                 | Unknown/Not available           | 404   |
>                 | User busy                       | 486   |
>                 | No reply                        | 408   |
>                 | Unconditional                   | 302   |
>                 | Deflection during alerting      | 487   |
>                 | Deflection immediate response   | 480   |
>                 | Mobile subscriber not reachable | 503   |
>                 +---------------------------------+-------+

> [MM2] OK, so the following hi-entry (index=1.3.1) must contain a
> cause URI parameter set to "cause=408" (removed from the draft-01
> version)

As I understand them, the suggestions are:

    The target sip:carol@example.com (index 1.2) is generated from
    sip:bob@example.com by "deflection immediate response", and so that
    URI should be augmented with the parameter cause=480.  I assume that
    parameter is carried into the hi-entry 1.2.1.

(It is not clear to me whether RFC 4458 applies exactly here, since
the deflection is not directly to voicemail but rather to Carol.
However, it seems that the intention of RFC 4458 is to describe how to
mark deflections to targets other than VM as well.  Probably
comparison with PSTN operation is the way to resolve this question.)

    The target sip:vm@example.com (index 1.3) is generated from
    sip:bob@example.com due to timeout of the INVITE to
    sip:carol@192.0.2.4 (index 1.2.1).  So hi-entry 1.2.1 gets
    "?Reason=SIP%3Bcause%3D408" and hi-entry 1.3 gets a URI parameter
    ";cause=408".  I assume that parameter is carried into the hi-entry
    1.3.1.

If I haven't made any mistakes, the marked changes need to be made:

  History-Info: <sip:bob@example.com>;index=1
  History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302>;\
                     index=1.1;rc=1
  History-Info: <sip:carol@example.com;cause=480>;index=1.2;mp=1
                                      ^^^^^^^^^^
  History-Info: <sip:carol@192.0.2.4;cause=480?Reason=SIP%3Bcause%3D408>;\
                                    ^^^^^^^^^^                      ^^^
                     index=1.2.1;rc=1.2
  History-Info: <sip:vm@example.com;\
                     target=sip:bob%40example.com;\
                                                 ^^
 		     cause=408>;\
                     ^^^^^^^^^
                     index=1.3;mp=1
  History-Info: <sip:vm@192.0.2.6;\
                     target=sip:bob%40example.com;\
                                                 ^^
 		     cause=408>;\
                     ^^^^^^^^^
                     index=1.3.1;rc=1.3

These changes need to be made to these hi-entries wherever they appear
in the example, except the "Reason=SIP%3Bcause%3D408" for 1.2.1 only
appears in F6 and F7.

Also, in F3, the Contact header becomes:

  Contact: <sip:carol@example.com;cause=480>;mp=1

Additionally, the last paragraph of the description of 3.6 seems to
have a damaged sentence:

   Furthermore it is the proxy forwarding the call to VMS that
   determines the target of the voicemail, it is the proxy that sets the
   target of voicemail which is also the entity that utilizes RFC4244bis
   to find the target which is usually based on local policy installed
   by the user or an administrator.

It might be helpful to add this note at F4:

  F4 INVITE Example.com -> Carol

  Because the call was forwarded from Bob to Carol due to "deflection
  immediate response", the URI parameter "cause=480" was added to
  Carol's AOR per [RFC448].  That parameter was carried through to
  Carol's contact URI during internal retargeting.

  INVITE sip:carol@192.0.2.4 SIP/2.0
  ...

Dale

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Subject: [sipcore] draft-ietf-sipcore-rfc4244bis-callflows-02: "Reason" placement
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In draft-ietf-sipcore-rfc4244bis-callflows-02, I find the following
inconsistency.  Could people please advise me what the correct usage
is?

When a non-2xx final response is received by a proxy (or synthesized,
in the case of timeouts), the response code is to be placed in the
Reason header value in an hi-entry.  It turns out that there are two
plausible choices:  One place is the hi-entry corresponding to the
INVITE that was sent by the proxy and which matches the response
received, the "next hop" location.  The other place is the hi-entry
which is the address of the UA which generated the response, the
"leaf" location.

draft-ietf-sipcore-rfc4244bis-callflows-02 is not rich in failure
responses, but there are these examples, with the corresponding
hi-entries for the failed fork:

   3.1.  Sequentially Forking (History-Info in Response)

   History-Info: <sip:office@example.com>;index=1.2;mp=1
   History-Info: <sip:office@192.0.2.5?Reason=SIP%3Bcause%3D408>;\
                 index=1.2.1;index=1.2.1;rc=1.2

   3.6.  PBX Voicemail Example

   History-Info: <sip:carol@example.com>;index=1.2;mp=1
   History-Info: <sip:carol@192.0.2.4?Reason=SIP%3Bcause%3D408>;\
		      index=1.2.1;rc=1.2

   3.7.  Consumer Voicemail Example

   History-Info: <sip:carol@example.com?Reason=SIP%3Bcause%3D408>;\
                      index=1.2;mp=1
   History-Info: <sip:carol@192.0.2.4>;index=1.2.1;rc=1.2

As you can see, these are inconsistent.

My belief (which may be faulty) is that the "Reason" goes in hi-entry
corresponding to the INVITE that the proxy sent, because that's what
the proxy can easily determine.  Trying to find the right "leaf"
hi-entry to attach the Reason to would be quite complicated in some
forking cases.

What is the right answer?

Dale

From marianne.mohali@orange.com  Thu Feb 21 00:52:01 2013
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From: <marianne.mohali@orange.com>
To: Shida Schubert <shida@ntt-at.com>
Thread-Topic: [sipcore] Verify draft-ietf-sipcore-rfc4244bis-callflows-02.txt
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Subject: Re: [sipcore] Verify draft-ietf-sipcore-rfc4244bis-callflows-02.txt
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Hi Shida,

My answers inline [MM2]

Marianne

> -----Message d'origine-----
> De=A0: Shida Schubert [mailto:shida@ntt-at.com]
> Envoy=E9=A0: mardi 19 f=E9vrier 2013 23:17
> =C0=A0: MOHALI Marianne OLNC/OLN
> Cc=A0: Paul Kyzivat; R.Jesske@telekom.de; Laura Liess; Brett Tate; Dale R.
> Worley; SIPCORE
> Objet=A0: Re: [sipcore] Verify draft-ietf-sipcore-rfc4244bis-callflows-
> 02.txt
>=20
>=20
> Hi Marianne;
>=20
>  Thank you for further comments.. My comments inline..
>=20
> On Feb 14, 2013, at 5:11 AM, <marianne.mohali@orange.com>
> <marianne.mohali@orange.com> wrote:
>=20
> > Hello,
> >
> > Most of my comments have been perfectly addressed but I have still
> some concerns with the following changes:
> >
> > MM#1 =A73.6 in F3: there is an "mp" tag in a Contact header; I guess
> it's a mistake and it should be a History-Info header (as in F4).
>=20
>  This is intentional. Per section  4.2.1 of RFC 4244bis regarding
> Redirect Server behavior, redirect server would indicate the method a
> redirect-to contact was determined.
>=20
[MM2] ok, understood




> >
> > MM#2 =A73.6 in F6: The History-Info header is
> > History-Info: <sip:bob@example.com>;index=3D1 			[MM] ok [/MM]
> > History-Info: <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302>;\
> >      		   index=3D1.1;rc=3D1 				[MM] ok [/MM]
> > History-Info: <sip:carol@example.com>;index=3D1.2;mp=3D1 	[MM] ok [/MM]
> > History-Info: <sip:carol@192.0.2.4?Reason=3DSIP%3Bcause%3D480>;\
> >                     index=3D1.2.1;rc=3D1.2 				[MM]
> nok, it is a cause URI parameter that should be set to "480" (call
> deflection) following RFC4458. The Reason header is escaped in the SIP
> URI only when it is a real received respond (it is the case with the
> 302) [/MM]
> > History-Info: <sip:vm@example.com;\
> >                     target=3Dsip:bob%40example.com>;\
> >                     index=3D1.3;mp=3D1 				[MM] nok,
> The hi-entry with the voicemail should be a copy of the R-URI so with
> the cause=3D480 URI parameter. Additionally, the vm is finally reached
> because Carol does not reply, so the no reply reason (cause=3D408) should
> also be reflected in a cause URI parameter following RFC4458. Hence,
> the hi-entry should be History-Info:
> <sip:vm@example.com;target=3Dsip:bob%40example.com;cause=3D480;cause=3D40=
8>;
> index=3D1.3;mp=3D1  =3D> Oops, I think we have a problem... This is due to
> the fact that at the same time we have successive forwarding (1 after
> the other) with Carol at the second-to-last position and we try to join
> B's vm (1st diverting user). IMO, we should add some text stating that
> depending on which voicemail is reached (Bob or Carol), the call
> forwarding reason (Bob or Carol's diverting reason) in the cause URI
> parameter should be set accordingly (to be consistent). Here the target
> vm is Bob so only cause=3D480 should be used and Carol's reason for not
> answering the call is lost. [/MM]
> > [MM]If someone have a better/other idea, please don't hesitate...
> > [/MM]
> > History-Info: <sip:vm@192.0.2.6;\
> >                     target=3Dsip:bob%40example.com>;\
> >                     index=3D1.3.1;rc=3D1.3 				[MM]
> nok, same comment
> >
> > But IMO it should be:
> >
> > History-Info: <sip:bob@example.com>;index=3D1
> > History-Info: <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302>;\
> >                     index=3D1.1;rc=3D1
> > History-Info: <sip:carol@example.com>;index=3D1.2;mp=3D1
> > History-Info: <sip:carol@192.0.2.4;cause=3D480>;\
> >                     index=3D1.2.1;rc=3D1.2
> > History-Info: <sip:vm@example.com;\
> >                     target=3Dsip:bob%40example.com;cause=3D480>;\
> >                     index=3D1.3;mp=3D1
> > History-Info: <sip:vm@192.0.2.6;\
> >                     target=3Dsip:bob%40example.com;cause=3D480>;\
> >                     index=3D1.3.1;rc=3D1.3
>=20
>  Ok. Fixed.
>=20

[MM2] After discussion with Dale, I'm not sure for the last hi-entry reason=
 value if it should be 408 (for the no reply from Carol) or 480 (for the in=
itial call deflection from bob). Indeed, in this last proposal, we lose the=
 no reply reason from Carol so set 408 could give this information but on t=
he other hand because there is mp=3D1 in the second-to-last hi-entry referi=
ng to bob whom reason is 480...=20
Finally, I think 480 is correct and the 2-last reasons should be 408 (no re=
ply) only if mp was=3D1.2. Do you agree?



> >
> > MM#3 =A73.7 in F4 and followings:
> > In the hi-entry the reason-text has been added.
> > I have 2 comments:
> > 1. why adding the reason-text? (and why not ;)=3D> ok for me 2. if the
> > reason-text is added, it should be "escaped" in the same way as the
> cause parameter.
> > So that, the hi-entry:
> > History-Info: <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302\
> >                         ;text=3D"Moved Temporarily">\
> >                 ;index=3D1.1;rc=3D1
> >
> > Should be:
> > History-Info: <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302\
> >                         %3Btext%3D"Moved Temporarily">\
> >                 ;index=3D1.1;rc=3D1
>=20
>  Good catch. Thanks
[MM2] Once again, check Dale's response to me because remaining " and =3D s=
eem also wrong. I didn't do the whole escaping.=20=20=20




>=20
>  Shida
>=20
> >
> > Here is my feedback.
> >
> > Best Regards,
> > Marianne
> >
> > -----Message d'origine-----
> > De : sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] De la
> > part de Paul Kyzivat Envoy=E9 : vendredi 1 f=E9vrier 2013 23:37 =C0 :
> > R.Jesske@telekom.de; Laura Liess; MOHALI Marianne OLNC/OLN; Brett
> > Tate; Dale R. Worley Cc : SIPCORE Objet : [sipcore] Verify
> > draft-ietf-sipcore-rfc4244bis-callflows-02.txt
> >
> > A new version of 4244bis-callflows has been published, to address the
> comments we received in the WGLC review at the end of last year.
> >
> > I've sent this *To* the people who provided comments.
> > I would appreciate your verifying that your comments were properly
> > addressed. (Please comment back one way or the other.)
> >
> > This is *not* a new WGLC. So please don't bring up *new* issues now.
> We are past that. I just want to verify that the comments already
> raised have been handled. We are holding the bis to publish
> concurrently with the callflows, so is the last thing holding them up.
> >
> > 	Thanks,
> > 	Paul
> >
> >
> > -------- Original Message --------
> > Subject: [sipcore] I-D Action:
> > draft-ietf-sipcore-rfc4244bis-callflows-02.txt
> > Date: Tue, 29 Jan 2013 12:49:12 -0800
> > From: internet-drafts@ietf.org
> > To: i-d-announce@ietf.org
> > CC: sipcore@ietf.org
> >
> >
> > A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> >  This draft is a work item of the Session Initiation Protocol Core
> Working Group of the IETF.
> >
> > 	Title           : Session Initiation Protocol (SIP) History-Info
> Header
> > Call Flow Examples
> > 	Author(s)       : Mary Barnes
> >                           Francois Audet
> >                           Shida Schubert
> >                           Detecon International Gmbh
> >                           Christer Holmberg
> > 	Filename        : draft-ietf-sipcore-rfc4244bis-callflows-02.txt
> > 	Pages           : 46
> > 	Date            : 2013-01-29
> >
> > Abstract:
> >    This document describes use cases and documents call flows which
> >    require the History-Info header field to capture the Request-URIs
> as
> >    a Session Initiation Protocol (SIP) Request is retargeted.  The
> use
> >    cases are described along with the corresponding call flow
> diagrams
> >    and messaging details.
> >
> >
> > The IETF datatracker status page for this draft is:
> > https://datatracker.ietf.org/doc/draft-ietf-sipcore-rfc4244bis-
> callflo
> > ws
> >
> > There's also a htmlized version available at:
> > http://tools.ietf.org/html/draft-ietf-sipcore-rfc4244bis-callflows-02
> >
> > A diff from the previous version is available at:
> > http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-sipcore-rfc4244bis-
> callflo
> > ws-02
> >
> >
> > Internet-Drafts are also available by anonymous FTP at:
> > ftp://ftp.ietf.org/internet-drafts/
> >
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
> >
> >
> >
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
> >
> >
> ______________________________________________________________________
> > ___________________________________________________
> >
> > Ce message et ses pieces jointes peuvent contenir des informations
> > confidentielles ou privilegiees et ne doivent donc pas etre diffuses,
> > exploites ou copies sans autorisation. Si vous avez recu ce message
> > par erreur, veuillez le signaler a l'expediteur et le detruire ainsi
> que les pieces jointes. Les messages electroniques etant susceptibles
> d'alteration, France Telecom - Orange decline toute responsabilite si
> ce message a ete altere, deforme ou falsifie. Merci.
> >
> > This message and its attachments may contain confidential or
> > privileged information that may be protected by law; they should not
> be distributed, used or copied without authorisation.
> > If you have received this email in error, please notify the sender
> and delete this message and its attachments.
> > As emails may be altered, France Telecom - Orange is not liable for
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> > Thank you.
> >
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore


___________________________________________________________________________=
______________________________________________

Ce message et ses pieces jointes peuvent contenir des informations confiden=
tielles ou privilegiees et ne doivent donc
pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu=
 ce message par erreur, veuillez le signaler
a l'expediteur et le detruire ainsi que les pieces jointes. Les messages el=
ectroniques etant susceptibles d'alteration,
France Telecom - Orange decline toute responsabilite si ce message a ete al=
tere, deforme ou falsifie. Merci.

This message and its attachments may contain confidential or privileged inf=
ormation that may be protected by law;
they should not be distributed, used or copied without authorisation.
If you have received this email in error, please notify the sender and dele=
te this message and its attachments.
As emails may be altered, France Telecom - Orange is not liable for message=
s that have been modified, changed or falsified.
Thank you.


From shinji.okumura@softfront.jp  Thu Feb 21 22:20:47 2013
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Subject: Re: [sipcore] I-D Action: draft-ietf-sipcore-rfc4244bis-callflows-02.txt
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Hi, all.

I'm interested in these drafts, but I'm confused a little.

draft-ietf-sipcore-rfc4244bis-callflows-02.txt Page 34
3.7.  Consumer Voicemail Example

In this scenario, I assume another indexing for last two History-Info headers.

[original]
   F6 INVITE Example.com -> VM

   INVITE sip:vm@192.0.2.6;target=sip:carol%40example.com SIP/2.0
   Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4bKbbg4
   Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
   Max-Forward: 67
   From: Alice <sip:alice@example.com>;tag=kkaz-
   To: Bob <sip:bob@example.com>
   Supported: histinfo
   Call-Id: 12345600@example.com
   CSeq: 1 INVITE
   History-Info: <sip:bob@example.com>;index=1
   History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302\
                      ;text="Moved Temporarily">;index=1.1;rc=1
   History-Info: <sip:carol@example.com?Reason=SIP%3Bcause%3D408>;\
                      index=1.2;mp=1
   History-Info: <sip:carol@192.0.2.4>;index=1.2.1;rc=1.2
   History-Info: <sip:vm@example.com;target=sip:carol%40example.com>;\
                      index=1.3;mp=1.2
   History-Info: <sip:vm@192.0.2.5;target=sip:carol%40example.com>\
                      ;cause=408;index=1.3.1;rc=1.3
   Contact: Alice <sip:alice@192.0.2.3>
   Content-Type: application/sdp
   Content-Length: <appropriate value>


[my indexing]
   INVITE sip:vm@192.0.2.6;target=sip:carol%40example.com;cause=408 SIP/2.0
   <snip/>
   History-Info: <sip:vm@example.com;target=sip:carol%40example.com;cause=408>;\
                      index=1.2.2;mp=1.2
   History-Info: <sip:vm@192.0.2.5;target=sip:carol%40example.com;cause=408>\
                      ;index=1.2.2.1;rc=1.2.2

Is my indexing against this specifications?

Regards,
Shinji.

internet-drafts@ietf.org
Tue, 29 Jan 2013 12:49:12 -0800
>
>A New Internet-Draft is available from the on-line Internet-Drafts directories.
> This draft is a work item of the Session Initiation Protocol Core Working Group of the IETF.
>
>	Title           : Session Initiation Protocol (SIP) History-Info Header Call Flow Examples
>	Author(s)       : Mary Barnes
>                          Francois Audet
>                          Shida Schubert
>                          Detecon International Gmbh
>                          Christer Holmberg
>	Filename        : draft-ietf-sipcore-rfc4244bis-callflows-02.txt
>	Pages           : 46
>	Date            : 2013-01-29
>
>Abstract:
>   This document describes use cases and documents call flows which
>   require the History-Info header field to capture the Request-URIs as
>   a Session Initiation Protocol (SIP) Request is retargeted.  The use
>   cases are described along with the corresponding call flow diagrams
>   and messaging details.
>
>
>The IETF datatracker status page for this draft is:
>https://datatracker.ietf.org/doc/draft-ietf-sipcore-rfc4244bis-callflows
>
>There's also a htmlized version available at:
>http://tools.ietf.org/html/draft-ietf-sipcore-rfc4244bis-callflows-02
>
>A diff from the previous version is available at:
>http://www.ietf.org/rfcdiff?url2=draft-ietf-sipcore-rfc4244bis-callflows-02
>
>
>Internet-Drafts are also available by anonymous FTP at:
>ftp://ftp.ietf.org/internet-drafts/

From shinji.okumura@softfront.jp  Mon Feb 25 02:03:00 2013
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From: OKUMURA Shinji <shinji.okumura@softfront.jp>
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Cc: "Dale R. Worley" <worley@ariadne.com>
Subject: Re: [sipcore] draft-ietf-sipcore-rfc4244bis-callflows-02: "Reason"placement
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Hi, Dale.

worley@ariadne.com (Dale R. Worley)
Wed, 20 Feb 2013 15:49:44 -0500 (EST)
>In draft-ietf-sipcore-rfc4244bis-callflows-02, I find the following
>inconsistency.  Could people please advise me what the correct usage
>is?
>
>When a non-2xx final response is received by a proxy (or synthesized,
>in the case of timeouts), the response code is to be placed in the
>Reason header value in an hi-entry.  It turns out that there are two
>plausible choices:  One place is the hi-entry corresponding to the
>INVITE that was sent by the proxy and which matches the response
>received, the "next hop" location.  The other place is the hi-entry
>which is the address of the UA which generated the response, the
>"leaf" location.
>
>draft-ietf-sipcore-rfc4244bis-callflows-02 is not rich in failure
>responses, but there are these examples, with the corresponding
>hi-entries for the failed fork:
>
>   3.1.  Sequentially Forking (History-Info in Response)
>
>   History-Info: <sip:office@example.com>;index=1.2;mp=1
>   History-Info: <sip:office@192.0.2.5?Reason=SIP%3Bcause%3D408>;\
>                 index=1.2.1;index=1.2.1;rc=1.2
>
>   3.6.  PBX Voicemail Example
>
>   History-Info: <sip:carol@example.com>;index=1.2;mp=1
>   History-Info: <sip:carol@192.0.2.4?Reason=SIP%3Bcause%3D408>;\
>		      index=1.2.1;rc=1.2
>
>   3.7.  Consumer Voicemail Example
>
>   History-Info: <sip:carol@example.com?Reason=SIP%3Bcause%3D408>;\
>                      index=1.2;mp=1
>   History-Info: <sip:carol@192.0.2.4>;index=1.2.1;rc=1.2
>
>As you can see, these are inconsistent.
>
>My belief (which may be faulty) is that the "Reason" goes in hi-entry
>corresponding to the INVITE that the proxy sent, because that's what
>the proxy can easily determine.  Trying to find the right "leaf"
>hi-entry to attach the Reason to would be quite complicated in some
>forking cases.
>
>What is the right answer?

draft-ietf-sipcore-rfc4244bis-11
10.2.  Reason in the History-Info Header Field

   If the Reason header field is being added due to
   receipt of an explicit SIP response and the response contains any
   Reason header fields (see [RFC3326]), then the SIP entity MUST
   include the Reason header fields in the "headers" component of the
   hi-targeted-to-uri in the last hi-entry added to the cache, unless
   the hi-targeted-to-uri is a Tel-URI.  If the SIP response does not
   contain a Reason header field, the SIP entity MUST include a Reason
   header field, containing the SIP Response Code, in the "headers"
   component of the hi-targeted-to-uri in the last hi-entry added to the
   cache, unless the hi-targeted-to-uri is a Tel-URI.

according to this descriptions, SIP entity MUST include a Reason header
"in the last hi-entry added to the cache".
IIUC, it means "next hop" index=1.2.1.

Regards,
Shinji.

From worley@shell01.TheWorld.com  Mon Feb 25 08:14:03 2013
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From: worley@ariadne.com (Dale R. Worley)
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Subject: Re: [sipcore] Fwd: Updates to draft-ietf-sipcore-rfc4244bis-11, version 1.
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> From: Mary Barnes <mary.ietf.barnes@gmail.com>
> 
> This is a set of suggested changes from Dale based upon the comments
> he sent to the list previously:
> http://www.ietf.org/mail-archive/web/sipcore/current/msg05489.html
> 
> Responses are inline below [MB].   Based upon my responses, I do not
> believe any of the changes impact normative behavior (as intended)
> thus another WG and/or IETF LC should not be necessary, but I will
> leave that decision to the chairs/AD.

Just to get everyone oriented:  I've been discussing my issues with
Mary Barnes.  After clearing away a number of minor points and
misunderstandings, there are two significant questions.

One question is properly describing how hi-entries are ordered.  The
terms "chronological order" and "tree preorder" have been used in the
past, but neither is completely correct.  I believe that Mary and I
have agreed that "lexicographic order by index values" is completely
correct and unambiguous.  I've proposed some wording changes to
support this, and Mary is reviewing them.

The other question regards the details of adding "Reason" values to
hi-entries in certain messy circumstances and ensuring the draft is
unambiguous about how they are to be handled.  I've formulated
examples and descriptions, and sent them to Mary.

Dale

From rjsparks@nostrum.com  Mon Feb 25 13:18:10 2013
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Subject: Re: [sipcore] AD Review: draft-ietf-sipcore-sip-websocket-06.txt
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On 2/18/13 5:10 PM, Paul Kyzivat wrote:
> On 2/17/13 8:31 PM, IÃ±aki Baz Castillo wrote:
>
>>> * I'm ok proceeding with the IANA Considerations sections as they 
>>> currently
>>>    stand, but I wonder if instead of what's in 10.3 and 10.4, this 
>>> document
>>>    should create a new registry for the transport values. That way 
>>> someone
>>>    could search the registry for the strings "WS" or "WSS" and find 
>>> something
>>>    meaningful.
>>
>> I'm ok with what you suggest, but I don't think I can start such a new
>> registry. If you consider that the document requires a new registry
>> please let me know how to help with it.
>
> We just went through this by establishing a new registry for the 
> priority header field.
>
> But in that case, the draft that provoked the creation of the new 
> registry was in a different WG, and we thought it weird for that draft 
> to establish a new registry for a sip header field.
>
> In this case everything is within sipcore, so I don't see any reason 
> why we couldn't do everything within this draft.
>
> Robert, what do you think?
Yes, in this draft is the right place to create the registry if the 
working group thinks a registry is a good idea.

Is there anyone who thinks this would be a bad step?


> Paul


From rjsparks@nostrum.com  Mon Feb 25 13:38:51 2013
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Subject: Re: [sipcore] AD Review: draft-ietf-sipcore-sip-websocket-06.txt
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--------------080304050408010704020502
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On 2/25/13 3:18 PM, Robert Sparks wrote:
> On 2/18/13 5:10 PM, Paul Kyzivat wrote:
>> On 2/17/13 8:31 PM, IÃ±aki Baz Castillo wrote:
>>
>>>> * I'm ok proceeding with the IANA Considerations sections as they 
>>>> currently
>>>>    stand, but I wonder if instead of what's in 10.3 and 10.4, this 
>>>> document
>>>>    should create a new registry for the transport values. That way 
>>>> someone
>>>>    could search the registry for the strings "WS" or "WSS" and find 
>>>> something
>>>>    meaningful.
>>>
>>> I'm ok with what you suggest, but I don't think I can start such a new
>>> registry. If you consider that the document requires a new registry
>>> please let me know how to help with it.
>>
>> We just went through this by establishing a new registry for the 
>> priority header field.
>>
>> But in that case, the draft that provoked the creation of the new 
>> registry was in a different WG, and we thought it weird for that 
>> draft to establish a new registry for a sip header field.
>>
>> In this case everything is within sipcore, so I don't see any reason 
>> why we couldn't do everything within this draft.
>>
>> Robert, what do you think?
> Yes, in this draft is the right place to create the registry if the 
> working group thinks a registry is a good idea.
>
> Is there anyone who thinks this would be a bad step?
If people _do_ think this is the right thing to do, this is the text I 
think we would need. (Based on a quick steal from
the sipcore-priority document):


10.6 SIP Transports Registry

   This document adds a new registry, "SIP Transport".
   Its format and initial values are as shown in the following table:

                         +------------+-----------+
                         | Priority   | Reference |
                         +------------+-----------+
                         | UDP        | RFC 3261  |
                         | TCP        | RFC 3261  |
                         | TLS        | RFC 3261  |
                         | SCTP       | RFC 3261  |
                         | WS         | RFC XXXX  |
                         | WSS        | RFC XXXX  |
                         +------------+-----------+

    The policy for registration of values in this registry is "Standards Action",
    as that term is defined by [RFC5226].

      RFC Editor Note: Please replace XXXX of the RFC number assigned 
for this document and
      remove this note.
>
>
>> Paul
>


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<html>
  <head>
    <meta content="text/html; charset=UTF-8" http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    <div class="moz-cite-prefix">On 2/25/13 3:18 PM, Robert Sparks
      wrote:<br>
    </div>
    <blockquote cite="mid:512BD50B.1040709@nostrum.com" type="cite">On
      2/18/13 5:10 PM, Paul Kyzivat wrote:
      <br>
      <blockquote type="cite">On 2/17/13 8:31 PM, IÃ±aki Baz Castillo
        wrote:
        <br>
        <br>
        <blockquote type="cite">
          <blockquote type="cite">* I'm ok proceeding with the IANA
            Considerations sections as they currently
            <br>
            Â Â  stand, but I wonder if instead of what's in 10.3 and
            10.4, this document
            <br>
            Â Â  should create a new registry for the transport values.
            That way someone
            <br>
            Â Â  could search the registry for the strings "WS" or "WSS"
            and find something
            <br>
            Â Â  meaningful.
            <br>
          </blockquote>
          <br>
          I'm ok with what you suggest, but I don't think I can start
          such a new
          <br>
          registry. If you consider that the document requires a new
          registry
          <br>
          please let me know how to help with it.
          <br>
        </blockquote>
        <br>
        We just went through this by establishing a new registry for the
        priority header field.
        <br>
        <br>
        But in that case, the draft that provoked the creation of the
        new registry was in a different WG, and we thought it weird for
        that draft to establish a new registry for a sip header field.
        <br>
        <br>
        In this case everything is within sipcore, so I don't see any
        reason why we couldn't do everything within this draft.
        <br>
        <br>
        Robert, what do you think?
        <br>
      </blockquote>
      Yes, in this draft is the right place to create the registry if
      the working group thinks a registry is a good idea.
      <br>
      <br>
      Is there anyone who thinks this would be a bad step?
      <br>
    </blockquote>
    If people _do_ think this is the right thing to do, this is the text
    I think we would need. (Based on a quick steal from<br>
    the sipcore-priority document):<br>
    <br>
    <br>
    10.6 SIP Transports Registry<br>
    <br>
    <meta http-equiv="content-type" content="text/html; charset=UTF-8">
    <pre>  This document adds a new registry, "SIP Transport".
  Its format and initial values are as shown in the following table:

                        +------------+-----------+
                        | Priority   | Reference |
                        +------------+-----------+
                        | UDP        | RFC 3261  |
                        | TCP        | RFC 3261  |
                        | TLS        | RFC 3261  |
                        | SCTP       | RFC 3261  |
                        | WS         | RFC XXXX  |
                        | WSS        | RFC XXXX  |
                      Â  +------------+-----------+

   The policy for registration of values in this registry is "Standards Action",
  Â as that term is defined by [RFC5226].</pre>
    Â Â Â Â  RFC Editor Note: Please replace XXXX of the RFC number assigned
    for this document and<br>
    Â Â Â Â  remove this note.<br>
    <blockquote cite="mid:512BD50B.1040709@nostrum.com" type="cite">
      <br>
      <br>
      <blockquote type="cite">Paul
        <br>
      </blockquote>
      <br>
    </blockquote>
    <br>
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Subject: Re: [sipcore] AD Review: draft-ietf-sipcore-sip-websocket-06.txt
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Looks like I missed 4168 - correction below:

On 2/25/13 3:38 PM, Robert Sparks wrote:
> On 2/25/13 3:18 PM, Robert Sparks wrote:
>> On 2/18/13 5:10 PM, Paul Kyzivat wrote:
>>> On 2/17/13 8:31 PM, Iñaki Baz Castillo wrote:
>>>
>>>>> * I'm ok proceeding with the IANA Considerations sections as they 
>>>>> currently
>>>>>    stand, but I wonder if instead of what's in 10.3 and 10.4, this 
>>>>> document
>>>>>    should create a new registry for the transport values. That way 
>>>>> someone
>>>>>    could search the registry for the strings "WS" or "WSS" and 
>>>>> find something
>>>>>    meaningful.
>>>>
>>>> I'm ok with what you suggest, but I don't think I can start such a new
>>>> registry. If you consider that the document requires a new registry
>>>> please let me know how to help with it.
>>>
>>> We just went through this by establishing a new registry for the 
>>> priority header field.
>>>
>>> But in that case, the draft that provoked the creation of the new 
>>> registry was in a different WG, and we thought it weird for that 
>>> draft to establish a new registry for a sip header field.
>>>
>>> In this case everything is within sipcore, so I don't see any reason 
>>> why we couldn't do everything within this draft.
>>>
>>> Robert, what do you think?
>> Yes, in this draft is the right place to create the registry if the 
>> working group thinks a registry is a good idea.
>>
>> Is there anyone who thinks this would be a bad step?
> If people _do_ think this is the right thing to do, this is the text I 
> think we would need. (Based on a quick steal from
> the sipcore-priority document):
>
>
> 10.6 SIP Transports Registry
>
>    This document adds a new registry, "SIP Transport".
>    Its format and initial values are as shown in the following table:

                         +------------+---------------------+
                         | Priority   | Reference           |
                         +------------+---------------------+
                         | UDP        | RFC 3261            |
                         | TCP        | RFC 3261            |
                         | TLS        | RFC 3261            |
                         | SCTP       | RFC 3261, RFC 4168  |
                         | TLS-SCTP   | RFC 4168            |
                         | WS         | RFC XXXX            |
                         | WSS        | RFC XXXX            |
                         +------------+---------------------+

>
>     The policy for registration of values in this registry is "Standards Action",
>     as that term is defined by [RFC5226].
>      RFC Editor Note: Please replace XXXX of the RFC number assigned 
> for this document and
>      remove this note.
>>
>>
>>> Paul
>>
>
>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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<html>
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    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
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    <div class="moz-cite-prefix">Looks like I missed 4168 - correction
      below:<br>
      <br>
      On 2/25/13 3:38 PM, Robert Sparks wrote:<br>
    </div>
    <blockquote cite="mid:512BD9E4.6020109@nostrum.com" type="cite">
      <meta content="text/html; charset=ISO-8859-1"
        http-equiv="Content-Type">
      <div class="moz-cite-prefix">On 2/25/13 3:18 PM, Robert Sparks
        wrote:<br>
      </div>
      <blockquote cite="mid:512BD50B.1040709@nostrum.com" type="cite">On
        2/18/13 5:10 PM, Paul Kyzivat wrote: <br>
        <blockquote type="cite">On 2/17/13 8:31 PM, I&ntilde;aki Baz Castillo
          wrote: <br>
          <br>
          <blockquote type="cite">
            <blockquote type="cite">* I'm ok proceeding with the IANA
              Considerations sections as they currently <br>
              &nbsp;&nbsp; stand, but I wonder if instead of what's in 10.3 and
              10.4, this document <br>
              &nbsp;&nbsp; should create a new registry for the transport values.
              That way someone <br>
              &nbsp;&nbsp; could search the registry for the strings "WS" or "WSS"
              and find something <br>
              &nbsp;&nbsp; meaningful. <br>
            </blockquote>
            <br>
            I'm ok with what you suggest, but I don't think I can start
            such a new <br>
            registry. If you consider that the document requires a new
            registry <br>
            please let me know how to help with it. <br>
          </blockquote>
          <br>
          We just went through this by establishing a new registry for
          the priority header field. <br>
          <br>
          But in that case, the draft that provoked the creation of the
          new registry was in a different WG, and we thought it weird
          for that draft to establish a new registry for a sip header
          field. <br>
          <br>
          In this case everything is within sipcore, so I don't see any
          reason why we couldn't do everything within this draft. <br>
          <br>
          Robert, what do you think? <br>
        </blockquote>
        Yes, in this draft is the right place to create the registry if
        the working group thinks a registry is a good idea. <br>
        <br>
        Is there anyone who thinks this would be a bad step? <br>
      </blockquote>
      If people _do_ think this is the right thing to do, this is the
      text I think we would need. (Based on a quick steal from<br>
      the sipcore-priority document):<br>
      <br>
      <br>
      10.6 SIP Transports Registry<br>
      <br>
      <meta http-equiv="content-type" content="text/html;
        charset=ISO-8859-1">
      <pre>  This document adds a new registry, "SIP Transport".
  Its format and initial values are as shown in the following table:
</pre>
    </blockquote>
    <br>
    <pre>                        +------------+---------------------+
                        | Priority   | Reference           |
                        +------------+---------------------+
                        | UDP        | RFC 3261            |
                        | TCP        | RFC 3261            |
                        | TLS        | RFC 3261            |
                        | SCTP       | RFC 3261, RFC 4168  |
                        | TLS-SCTP   | RFC 4168            |
                       &nbsp;| WS         | RFC XXXX            |
                        | WSS        | RFC XXXX            |
                      &nbsp; +------------+---------------------+</pre>
    <blockquote cite="mid:512BD9E4.6020109@nostrum.com" type="cite">
      <pre>

   The policy for registration of values in this registry is "Standards Action",
  &nbsp;as that term is defined by [RFC5226].</pre>
      &nbsp;&nbsp;&nbsp;&nbsp; RFC Editor Note: Please replace XXXX of the RFC number
      assigned for this document and<br>
      &nbsp;&nbsp;&nbsp;&nbsp; remove this note.<br>
      <blockquote cite="mid:512BD50B.1040709@nostrum.com" type="cite"> <br>
        <br>
        <blockquote type="cite">Paul <br>
        </blockquote>
        <br>
      </blockquote>
      <br>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
sipcore mailing list
<a class="moz-txt-link-abbreviated" href="mailto:sipcore@ietf.org">sipcore@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/sipcore">https://www.ietf.org/mailman/listinfo/sipcore</a>
</pre>
    </blockquote>
    <br>
  </body>
</html>

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Hi Iñaki (and SIPCORE participants):

One thing from my initial review that I don't think I've seen a response 
to (nor do I see how the draft revision addressed it) -
apologies if we had discussed this and I've forgotten:


On 11/27/12 4:35 PM, Robert Sparks wrote:
>
> * Should there be some additional discussion of what an element should 
> do if
>   someone clicks on "sip:foo.example.com;transport=ws" or receives 
> that in a
>   contact in a 302 or in a Refer-To: header field?
>   Taken to an extreme, what's a proxy that understands sip over 
> websockets
>   supposed to do if it gets a message containing Route header fields 
> (think
>   preloaded routes) where the next route value is for a transport=ws 
> hop that
>   a connection doesn't yet exist for?
>
RjS


From keith.drage@alcatel-lucent.com  Mon Feb 25 16:25:46 2013
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Robert Sparks <rjsparks@nostrum.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Date: Tue, 26 Feb 2013 01:25:37 +0100
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The IANA considerations for the new registry needs to indicate it is groupe=
d with the other SIP registries.

I see Robert has already spotted RFC 4168.

Keith

________________________________
From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf =
Of Robert Sparks
Sent: 25 February 2013 21:39
To: Paul Kyzivat
Cc: sipcore@ietf.org
Subject: Re: [sipcore] AD Review: draft-ietf-sipcore-sip-websocket-06.txt

On 2/25/13 3:18 PM, Robert Sparks wrote:
On 2/18/13 5:10 PM, Paul Kyzivat wrote:
On 2/17/13 8:31 PM, I=F1aki Baz Castillo wrote:

* I'm ok proceeding with the IANA Considerations sections as they currently
   stand, but I wonder if instead of what's in 10.3 and 10.4, this document
   should create a new registry for the transport values. That way someone
   could search the registry for the strings "WS" or "WSS" and find somethi=
ng
   meaningful.

I'm ok with what you suggest, but I don't think I can start such a new
registry. If you consider that the document requires a new registry
please let me know how to help with it.

We just went through this by establishing a new registry for the priority h=
eader field.

But in that case, the draft that provoked the creation of the new registry =
was in a different WG, and we thought it weird for that draft to establish =
a new registry for a sip header field.

In this case everything is within sipcore, so I don't see any reason why we=
 couldn't do everything within this draft.

Robert, what do you think?
Yes, in this draft is the right place to create the registry if the working=
 group thinks a registry is a good idea.

Is there anyone who thinks this would be a bad step?
If people _do_ think this is the right thing to do, this is the text I thin=
k we would need. (Based on a quick steal from
the sipcore-priority document):


10.6 SIP Transports Registry

  This document adds a new registry, "SIP Transport".

  Its format and initial values are as shown in the following table:



                        +------------+-----------+

                        | Priority   | Reference |

                        +------------+-----------+

                        | UDP        | RFC 3261  |

                        | TCP        | RFC 3261  |

                        | TLS        | RFC 3261  |

                        | SCTP       | RFC 3261  |

                        | WS         | RFC XXXX  |

                        | WSS        | RFC XXXX  |

                        +------------+-----------+



   The policy for registration of values in this registry is "Standards Act=
ion",

   as that term is defined by [RFC5226].
     RFC Editor Note: Please replace XXXX of the RFC number assigned for th=
is document and
     remove this note.


Paul



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<body bgcolor=3Dwhite lang=3DEN-GB link=3Dblue vlink=3D"#606420">

<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>The IANA considerations for the new
registry needs to indicate it is grouped with the other SIP registries.<o:p=
></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>I see Robert has already spotted RFC 4=
168.<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'>Keith<o:p></o:p></span></font></p>

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

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0cm 0cm 0cm =
4.0pt'>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
color=3Dblack face=3D"Times New Roman"><span lang=3DEN-US style=3D'font-siz=
e:12.0pt;
color:windowtext'>

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

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

<p class=3DMsoNormal><b><font size=3D2 color=3Dblack face=3DTahoma><span la=
ng=3DEN-US
style=3D'font-size:10.0pt;font-family:Tahoma;color:windowtext;font-weight:b=
old'>From:</span></font></b><font
size=3D2 color=3Dblack face=3DTahoma><span lang=3DEN-US style=3D'font-size:=
10.0pt;
font-family:Tahoma;color:windowtext'> sipcore-bounces@ietf.org
[mailto:sipcore-bounces@ietf.org] <b><span style=3D'font-weight:bold'>On Be=
half
Of </span></b>Robert Sparks<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> 25 February 2013 21:39=
<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Paul Kyzivat<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> sipcore@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [sipcore] AD Re=
view:
draft-ietf-sipcore-sip-websocket-06.txt</span></font><font color=3Dblack><s=
pan
lang=3DEN-US style=3D'color:windowtext'><o:p></o:p></span></font></p>

</div>

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

<div>

<p class=3DMsoNormal><font size=3D3 color=3Dblack face=3D"Times New Roman">=
<span
style=3D'font-size:12.0pt'>On <st1:date Year=3D"13" Day=3D"25" Month=3D"2" =
ls=3D"trans"
w:st=3D"on">2/25/13</st1:date> <st1:time Minute=3D"18" Hour=3D"15" w:st=3D"=
on">3:18 PM</st1:time>,
Robert Sparks wrote:<o:p></o:p></span></font></p>

</div>

<blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'
cite=3D"mid:512BD50B.1040709@nostrum.com" type=3Dcite>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D3 color=3D=
black
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>On <st1:date Year=
=3D"13"
Day=3D"18" Month=3D"2" ls=3D"trans" w:st=3D"on">2/18/13</st1:date> <st1:tim=
e Minute=3D"10"
Hour=3D"17" w:st=3D"on">5:10 PM</st1:time>, Paul Kyzivat wrote: <o:p></o:p>=
</span></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D3 color=3D=
black
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>On <st1:date Year=
=3D"13"
Day=3D"17" Month=3D"2" ls=3D"trans" w:st=3D"on">2/17/13</st1:date> <st1:tim=
e Minute=3D"31"
Hour=3D"20" w:st=3D"on">8:31 PM</st1:time>, I=F1aki Baz Castillo wrote: <br=
>
<br>
<o:p></o:p></span></font></p>

<blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt' type=3Dcite>

<p class=3DMsoNormal><font size=3D3 color=3Dblack face=3D"Times New Roman">=
<span
style=3D'font-size:12.0pt'>* I'm ok proceeding with the IANA Considerations
sections as they currently <br>
&nbsp;&nbsp; stand, but I wonder if instead of what's in 10.3 and 10.4, thi=
s
document <br>
&nbsp;&nbsp; should create a new registry for the transport values. That wa=
y
someone <br>
&nbsp;&nbsp; could search the registry for the strings &quot;WS&quot; or
&quot;WSS&quot; and find something <br>
&nbsp;&nbsp; meaningful. <o:p></o:p></span></font></p>

</blockquote>

<p class=3DMsoNormal><font size=3D3 color=3Dblack face=3D"Times New Roman">=
<span
style=3D'font-size:12.0pt'><br>
I'm ok with what you suggest, but I don't think I can start such a new <br>
registry. If you consider that the document requires a new registry <br>
please let me know how to help with it. <o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 color=3Dblack face=3D"Times New Roman">=
<span
style=3D'font-size:12.0pt'><br>
We just went through this by establishing a new registry for the priority
header field. <br>
<br>
But in that case, the draft that provoked the creation of the new registry =
was
in a different WG, and we thought it weird for that draft to establish a ne=
w
registry for a sip header field. <br>
<br>
In this case everything is within sipcore, so I don't see any reason why we
couldn't do everything within this draft. <br>
<br>
Robert, what do you think? <o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 color=3Dblack face=3D"Times New Roman">=
<span
style=3D'font-size:12.0pt'>Yes, in this draft is the right place to create =
the
registry if the working group thinks a registry is a good idea. <br>
<br>
Is there anyone who thinks this would be a bad step? <o:p></o:p></span></fo=
nt></p>

</blockquote>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D3 color=3D=
black
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>If people _do_ th=
ink this
is the right thing to do, this is the text I think we would need. (Based on=
 a
quick steal from<br>
the sipcore-priority document):<br>
<br>
<br>
10.6 SIP Transports Registry<o:p></o:p></span></font></p>

<pre><font size=3D2 color=3Dblack face=3D"Courier New"><span style=3D'font-=
size:10.0pt'>&nbsp; This document adds a new registry, &quot;SIP Transport&=
quot;.<o:p></o:p></span></font></pre><pre><font
size=3D2 color=3Dblack face=3D"Courier New"><span style=3D'font-size:10.0pt=
'>&nbsp; Its format and initial values are as shown in the following table:=
<o:p></o:p></span></font></pre><pre><font
size=3D2 color=3Dblack face=3D"Courier New"><span style=3D'font-size:10.0pt=
'><o:p>&nbsp;</o:p></span></font></pre><pre><font
size=3D2 color=3Dblack face=3D"Courier New"><span style=3D'font-size:10.0pt=
'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; +--------=
----+-----------+<o:p></o:p></span></font></pre><pre><font
size=3D2 color=3Dblack face=3D"Courier New"><span style=3D'font-size:10.0pt=
'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | Priorit=
y&nbsp;&nbsp; | Reference |<o:p></o:p></span></font></pre><pre><font
size=3D2 color=3Dblack face=3D"Courier New"><span style=3D'font-size:10.0pt=
'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; +--------=
----+-----------+<o:p></o:p></span></font></pre><pre><font
size=3D2 color=3Dblack face=3D"Courier New"><span style=3D'font-size:10.0pt=
'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | UDP&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | RFC 3261&nbsp; |<o:p></o:p></span>=
</font></pre><pre><font
size=3D2 color=3Dblack face=3D"Courier New"><span style=3D'font-size:10.0pt=
'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | <st1:st=
ockticker
w:st=3D"on">TCP</st1:stockticker>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
 | RFC 3261&nbsp; |<o:p></o:p></span></font></pre><pre><font
size=3D2 color=3Dblack face=3D"Courier New"><span style=3D'font-size:10.0pt=
'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | <st1:st=
ockticker
w:st=3D"on">TLS</st1:stockticker>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
 | RFC 3261&nbsp; |<o:p></o:p></span></font></pre><pre><font
size=3D2 color=3Dblack face=3D"Courier New"><span style=3D'font-size:10.0pt=
'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | SCTP&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | RFC 3261&nbsp; |<o:p></o:p></span></fon=
t></pre><pre><font
size=3D2 color=3Dblack face=3D"Courier New"><span style=3D'font-size:10.0pt=
'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | WS&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | RFC XXXX&nbsp; |<o:p></o:p></=
span></font></pre><pre><font
size=3D2 color=3Dblack face=3D"Courier New"><span style=3D'font-size:10.0pt=
'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | WSS&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | RFC XXXX&nbsp; |<o:p></o:p></span>=
</font></pre><pre><font
size=3D2 color=3Dblack face=3D"Courier New"><span style=3D'font-size:10.0pt=
'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &nbsp; +------------+=
-----------+<o:p></o:p></span></font></pre><pre><font
size=3D2 color=3Dblack face=3D"Courier New"><span style=3D'font-size:10.0pt=
'><o:p>&nbsp;</o:p></span></font></pre><pre><font
size=3D2 color=3Dblack face=3D"Courier New"><span style=3D'font-size:10.0pt=
'>&nbsp;&nbsp; The policy for registration of values in this registry is &q=
uot;Standards Action&quot;,<o:p></o:p></span></font></pre><pre><font
size=3D2 color=3Dblack face=3D"Courier New"><span style=3D'font-size:10.0pt=
'>&nbsp; &nbsp;as that term is defined by [RFC5226].<o:p></o:p></span></fon=
t></pre>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D3 color=3D=
black
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>&nbsp;&nbsp;&nbsp=
;&nbsp;
RFC Editor Note: Please replace XXXX of the RFC number assigned for this
document and<br>
&nbsp;&nbsp;&nbsp;&nbsp; remove this note.<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D3 color=3D=
black
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'><br>
<br>
<o:p></o:p></span></font></p>

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

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

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

</div>

</div>

</body>

</html>

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From R.Jesske@telekom.de  Tue Feb 26 02:16:32 2013
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From: <R.Jesske@telekom.de>
To: <pkyzivat@alum.mit.edu>, <laura.liess.dt@googlemail.com>, <marianne.mohali@orange-ftgroup.com>, <brett@broadsoft.com>, <worley@ariadne.com>
Date: Tue, 26 Feb 2013 11:16:13 +0100
Thread-Topic: Verify draft-ietf-sipcore-rfc4244bis-callflows-02.txt
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Subject: Re: [sipcore] Verify draft-ietf-sipcore-rfc4244bis-callflows-02.txt
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Hi Paul,
I went through the draft and I'm happy with the status as it is.
So from my side no more open issues and all my comments are covered.

Thank you and Best Regards

Roland

> -----Urspr=FCngliche Nachricht-----
> Von: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
> Gesendet: Freitag, 1. Februar 2013 23:37
> An: Jesske, Roland; Laura Liess;
> marianne.mohali@orange-ftgroup.com; Brett Tate; Dale R. Worley
> Cc: SIPCORE
> Betreff: Verify draft-ietf-sipcore-rfc4244bis-callflows-02.txt
>
> A new version of 4244bis-callflows has been published, to
> address the comments we received in the WGLC review at the
> end of last year.
>
> I've sent this *To* the people who provided comments.
> I would appreciate your verifying that your comments were
> properly addressed. (Please comment back one way or the other.)
>
> This is *not* a new WGLC. So please don't bring up *new*
> issues now. We are past that. I just want to verify that the
> comments already raised have been handled. We are holding the
> bis to publish concurrently with the callflows, so is the
> last thing holding them up.
>
>       Thanks,
>       Paul
>
>
> -------- Original Message --------
> Subject: [sipcore] I-D Action:
> draft-ietf-sipcore-rfc4244bis-callflows-02.txt
> Date: Tue, 29 Jan 2013 12:49:12 -0800
> From: internet-drafts@ietf.org
> To: i-d-announce@ietf.org
> CC: sipcore@ietf.org
>
>
> A New Internet-Draft is available from the on-line
> Internet-Drafts directories.
>   This draft is a work item of the Session Initiation
> Protocol Core Working Group of the IETF.
>
>       Title           : Session Initiation Protocol (SIP)
> History-Info Header
> Call Flow Examples
>       Author(s)       : Mary Barnes
>                            Francois Audet
>                            Shida Schubert
>                            Detecon International Gmbh
>                            Christer Holmberg
>       Filename        : draft-ietf-sipcore-rfc4244bis-callflows-02.txt
>       Pages           : 46
>       Date            : 2013-01-29
>
> Abstract:
>     This document describes use cases and documents call flows which
>     require the History-Info header field to capture the
> Request-URIs as
>     a Session Initiation Protocol (SIP) Request is
> retargeted.  The use
>     cases are described along with the corresponding call
> flow diagrams
>     and messaging details.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-sipcore-rfc4244bis
> -callflows
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-sipcore-rfc4244bis-callflows-02
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-sipcore-rfc4244bis
> -callflows-02
>
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>
>
>
>

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 26 Feb 2013 11:16:14 +0100
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>, draft-ietf-sipcore-sip-websocket@tools.ietf.org
Subject: Re: [sipcore] AD Review: draft-ietf-sipcore-sip-websocket-06.txt
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Hi Robert, we are currently at MWC in Barcelona. We will address the
issues you and Paul have reported during next days.

Thanks a lot.



2013/2/25 Robert Sparks <rjsparks@nostrum.com>:
> Hi I=C3=B1aki (and SIPCORE participants):
>
> One thing from my initial review that I don't think I've seen a response =
to
> (nor do I see how the draft revision addressed it) -
> apologies if we had discussed this and I've forgotten:
>
>
>
> On 11/27/12 4:35 PM, Robert Sparks wrote:
>>
>>
>> * Should there be some additional discussion of what an element should d=
o
>> if
>>   someone clicks on "sip:foo.example.com;transport=3Dws" or receives tha=
t in
>> a
>>   contact in a 302 or in a Refer-To: header field?
>>   Taken to an extreme, what's a proxy that understands sip over websocke=
ts
>>   supposed to do if it gets a message containing Route header fields
>> (think
>>   preloaded routes) where the next route value is for a transport=3Dws h=
op
>> that
>>   a connection doesn't yet exist for?
>>
> RjS
>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore



--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From mary.ietf.barnes@gmail.com  Tue Feb 26 14:10:20 2013
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From: Mary Barnes <mary.ietf.barnes@gmail.com>
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Cc: Cullen Jennings <fluffy@cisco.com>, Richard Barnes <rbarnes@bbn.com>, SIPCORE <sipcore@ietf.org>, "rai-ads@tools.ietf.org" <rai-ads@tools.ietf.org>
Subject: [sipcore] Dispatching of draft-ivov-dispatch-trickle-ice-sip-00 and draft-ivov-dispatch-sdpfrag
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Hi all,

We had our review of the DISPATCH-86 topics with the ADs this morning.
 The decision was to dispatch these documents to the MMUSIC WG for
discussion as potential work items:
- draft-ivov-dispatch-trickle-ice-sip (currently proposes a new INFO
event package)
- draft-ivov-dispatch-sdpfrag (defines application/sdpfrag MIME media type)

It made more sense to dispatch to MMUSIC since they are related to
draft-ivov-mmusic-trickle-ice-00 which is currently under discussion
as a proposed MMUSIC WG document.  And, neither the SIP change nor RFC
6086 require that this sort of work be done in SIPCORE.

If you have any questions or comments, please let the ADs know ASAP.
I will be posting this decision along with the plans for the other
IETF-86 topics and agenda by tomorrow morning.

Regards,
Mary
DISPATCH WG co-chair

From worley@shell01.TheWorld.com  Tue Feb 26 19:23:04 2013
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From: worley@ariadne.com (Dale R. Worley)
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To: OKUMURA Shinji <shinji.okumura@softfront.jp>
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Cc: sipcore@ietf.org
Subject: Re: [sipcore] I-D Action:	draft-ietf-sipcore-rfc4244bis-callflows-02.txt
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> From: OKUMURA Shinji <shinji.okumura@softfront.jp>

> [original]
>    F6 INVITE Example.com -> VM
> 
>    INVITE sip:vm@192.0.2.6;target=sip:carol%40example.com SIP/2.0
>    Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4bKbbg4
>    Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
>    Max-Forward: 67
>    From: Alice <sip:alice@example.com>;tag=kkaz-
>    To: Bob <sip:bob@example.com>
>    Supported: histinfo
>    Call-Id: 12345600@example.com
>    CSeq: 1 INVITE
>    History-Info: <sip:bob@example.com>;index=1
>    History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302\
>                       ;text="Moved Temporarily">;index=1.1;rc=1
>    History-Info: <sip:carol@example.com?Reason=SIP%3Bcause%3D408>;\
>                       index=1.2;mp=1
>    History-Info: <sip:carol@192.0.2.4>;index=1.2.1;rc=1.2
>    History-Info: <sip:vm@example.com;target=sip:carol%40example.com>;\
>                       index=1.3;mp=1.2
>    History-Info: <sip:vm@192.0.2.5;target=sip:carol%40example.com>\
>                       ;cause=408;index=1.3.1;rc=1.3
>    Contact: Alice <sip:alice@192.0.2.3>
>    Content-Type: application/sdp
>    Content-Length: <appropriate value>

> [my indexing]
>    INVITE sip:vm@192.0.2.6;target=sip:carol%40example.com;cause=408 SIP/2.0
>    <snip/>
>    History-Info: <sip:vm@example.com;target=sip:carol%40example.com;cause=408>;\
>                       index=1.2.2;mp=1.2
>    History-Info: <sip:vm@192.0.2.5;target=sip:carol%40example.com;cause=408>\
>                       ;index=1.2.2.1;rc=1.2.2

I believe that you've found an error.

If the call was sent to *Bob's* voicemail, then the fork to VM would
be a child of the INVITE to sip:bob@example.com (index 1), so the
INVITE to voicemail would have index 1.3.

But in this case, the call was sent to *Carol's* voicemail, it is
(most likely) a child of the INVITE to sip:carol@example.com (index
1.2), and so the INVITE to voicemail would have index 1.2.2.

Dale

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Cc: sipcore@ietf.org
Subject: Re: [sipcore] draft-ietf-sipcore-rfc4244bis-callflows-02:	"Reason"placement
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> From: OKUMURA Shinji <shinji.okumura@softfront.jp>

[choosing one examle from several:]
> >   3.1.  Sequentially Forking (History-Info in Response)
> >
> >   History-Info: <sip:office@example.com>;index=1.2;mp=1
> >   History-Info: <sip:office@192.0.2.5?Reason=SIP%3Bcause%3D408>;\
> >                 index=1.2.1;index=1.2.1;rc=1.2

> draft-ietf-sipcore-rfc4244bis-11
> 10.2.  Reason in the History-Info Header Field
> 
>    If the Reason header field is being added due to
>    receipt of an explicit SIP response and the response contains any
>    Reason header fields (see [RFC3326]), then the SIP entity MUST
>    include the Reason header fields in the "headers" component of the
>    hi-targeted-to-uri in the last hi-entry added to the cache, unless
>    the hi-targeted-to-uri is a Tel-URI.  If the SIP response does not
>    contain a Reason header field, the SIP entity MUST include a Reason
>    header field, containing the SIP Response Code, in the "headers"
>    component of the hi-targeted-to-uri in the last hi-entry added to the
>    cache, unless the hi-targeted-to-uri is a Tel-URI.
> 
> according to this descriptions, SIP entity MUST include a Reason header
> "in the last hi-entry added to the cache".
> IIUC, it means "next hop" index=1.2.1.

Looking at the matter more closely, it is rather complex:

The text says "the last hi-entry added to the cache".  I believe that
it means "the hi-entry that was most recently added to the cache".

The processing described is in section 10.2.  The only location where
10.2 is referenced is in Step 2 of section 9.3.  Section 9.3 begins:

   9.3.  Receiving a Response with History-Info or Request Timeouts

   When a SIP entity receives a non-100 response or a request times out,
   the SIP entity performs the following steps:

   Step 1:  Add hi-entry to cache

      The SIP entity MUST add the hi-entry that was added to the request
      that received the non-100 response or timed out to the cache, if
      it was not already cached.  The hi-entry MUST be added to the
      cache in ascending order as indicated by the values in the hi-
      index parameters of the hi-entries (e.g., 1.2.1 comes after 1.2
      but before 1.2.2 or 1.3).

   Step 2:  Add Reason header field

      The SIP entity adds one or more Reason header fields to the hi-
      targeted-to-uri in the (newly) cached hi-entry reflecting the SIP
      response code in the non-100 response, per the procedures of
      Section 10.2.

As you can see, the point where section 10.2 is invoked is in Step 2,
and at that point the hi-entry that was most recently added to the
cache is the hi-entry specified in Step 1, namely, "the hi-entry that
was added to the request that received the ... response".  (hi-entries
taken from the response itself are added to the cache later, in Step
3.)

In this case, I believe there is a problem with the wording:  Step 1
does nothing because the specified hi-entry is already in the cache
due to reception of response F7 (in the callflow 3.1), and so the
specified hi-entry is not "added to the cache", but I believe it is
intended to be the hi-entry upon which Step 2 and section 10.2
operate.

Within that understanding, it seems to me that the Reason value should
be added to hi-entry 1.2.1, because that is the hi-entry that
corresponds to the INVITE that was sent from proxy "example.com" to
Office.

There is additional complexity because this is a case of "internal
retargeting":  It is as if the proxy sent "INVITE
sip:office@example.com" (index 1.2) to another proxy, which sent
"INVITE sip:office@192.0.2.5" (index 1.2.1).

Clearly (?), "example.com" should apply "Reason=...408..." to hi-entry
1.2.1, because that corresponds to the request that timed out.

But it seems to me that "example.com", in its role as the upstream
proxy which internally receives the 408 response from itself in its
role as the downstream proxy, should apply "Reason=...408..." to
hi-entry 1.2 as well.  (If the two proxies were separated, that is
what would happen.)

Dale

From shinji.okumura@softfront.jp  Wed Feb 27 00:22:56 2013
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Subject: Re: [sipcore] draft-ietf-sipcore-rfc4244bis-callflows-02:"Reason"placement
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Hi, Dale.

Thank you for your response, my comments inline.

worley@ariadne.com (Dale R. Worley)
Tue, 26 Feb 2013 22:22:01 -0500 (EST)
>> From: OKUMURA Shinji <shinji.okumura@softfront.jp>
>
>[choosing one examle from several:]
>> >   3.1.  Sequentially Forking (History-Info in Response)
>> >
>> >   History-Info: <sip:office@example.com>;index=1.2;mp=1
>> >   History-Info: <sip:office@192.0.2.5?Reason=SIP%3Bcause%3D408>;\
>> >                 index=1.2.1;index=1.2.1;rc=1.2
>
>> draft-ietf-sipcore-rfc4244bis-11
>> 10.2.  Reason in the History-Info Header Field
>> 
>>    If the Reason header field is being added due to
>>    receipt of an explicit SIP response and the response contains any
>>    Reason header fields (see [RFC3326]), then the SIP entity MUST
>>    include the Reason header fields in the "headers" component of the
>>    hi-targeted-to-uri in the last hi-entry added to the cache, unless
>>    the hi-targeted-to-uri is a Tel-URI.  If the SIP response does not
>>    contain a Reason header field, the SIP entity MUST include a Reason
>>    header field, containing the SIP Response Code, in the "headers"
>>    component of the hi-targeted-to-uri in the last hi-entry added to the
>>    cache, unless the hi-targeted-to-uri is a Tel-URI.
>> 
>> according to this descriptions, SIP entity MUST include a Reason header
>> "in the last hi-entry added to the cache".
>> IIUC, it means "next hop" index=1.2.1.
>
>Looking at the matter more closely, it is rather complex:
>
>The text says "the last hi-entry added to the cache".  I believe that
>it means "the hi-entry that was most recently added to the cache".
>
>The processing described is in section 10.2.  The only location where
>10.2 is referenced is in Step 2 of section 9.3.  Section 9.3 begins:
>
>   9.3.  Receiving a Response with History-Info or Request Timeouts
>
>   When a SIP entity receives a non-100 response or a request times out,
>   the SIP entity performs the following steps:
>
>   Step 1:  Add hi-entry to cache
>
>      The SIP entity MUST add the hi-entry that was added to the request
>      that received the non-100 response or timed out to the cache, if
>      it was not already cached.  The hi-entry MUST be added to the
>      cache in ascending order as indicated by the values in the hi-
>      index parameters of the hi-entries (e.g., 1.2.1 comes after 1.2
>      but before 1.2.2 or 1.3).
>
>   Step 2:  Add Reason header field
>
>      The SIP entity adds one or more Reason header fields to the hi-
>      targeted-to-uri in the (newly) cached hi-entry reflecting the SIP
>      response code in the non-100 response, per the procedures of
>      Section 10.2.
>
>As you can see, the point where section 10.2 is invoked is in Step 2,
>and at that point the hi-entry that was most recently added to the
>cache is the hi-entry specified in Step 1, namely, "the hi-entry that
>was added to the request that received the ... response".  (hi-entries
>taken from the response itself are added to the cache later, in Step
>3.)

At a time when the SIP entity adds one or more Reason header
fields, "the index of the last hi-entry added to the cache"
is 1.2.1, isn't it?

>In this case, I believe there is a problem with the wording:  Step 1
>does nothing because the specified hi-entry is already in the cache
>due to reception of response F7 (in the callflow 3.1), and so the
>specified hi-entry is not "added to the cache", but I believe it is
>intended to be the hi-entry upon which Step 2 and section 10.2
>operate.

aha!

At proxy.example.com,
  when                  to the cache
  -------------------------------------------------
  receiving INVITE(F1)  add index=1
  sending INVITE(F2)    add index=1.1
  receiving 302(F4)     include a Reason in index=1.1(last at the time)
  sending INVITE(F6)    add index=1.2
                        add index=1.2.1
  receiving 180(F7)     add nothig. but it is terrible, if 180 has index=1.2.1.x...
  time out(--)          include a Reason in index=1.2.1(last at the time)

therefore,
the SIP entity MUST include a Reason header field in the last
hi-entry "created by itself".

>Within that understanding, it seems to me that the Reason value should
>be added to hi-entry 1.2.1, because that is the hi-entry that
>corresponds to the INVITE that was sent from proxy "example.com" to
>Office.

I think so too.

>There is additional complexity because this is a case of "internal
>retargeting":  It is as if the proxy sent "INVITE
>sip:office@example.com" (index 1.2) to another proxy, which sent
>"INVITE sip:office@192.0.2.5" (index 1.2.1).
>
>Clearly (?), "example.com" should apply "Reason=...408..." to hi-entry
>1.2.1, because that corresponds to the request that timed out.
>
>But it seems to me that "example.com", in its role as the upstream
>proxy which internally receives the 408 response from itself in its
>role as the downstream proxy, should apply "Reason=...408..." to
>hi-entry 1.2 as well.  (If the two proxies were separated, that is
>what would happen.)

it seems fine that both hi-entries(1.2 and 1.2.1) have a Reason header.
The last one MUST include it, another one "should" do it.

Regards,
Shinji.

From shinji.okumura@softfront.jp  Wed Feb 27 00:22:56 2013
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Subject: Re: [sipcore] I-D Action:draft-ietf-sipcore-rfc4244bis-callflows-02.txt
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Hi, Dale.

Thank you for your response, my comments inline.

worley@ariadne.com (Dale R. Worley)
Tue, 26 Feb 2013 22:22:23 -0500 (EST)
>> From: OKUMURA Shinji <shinji.okumura@softfront.jp>
>
>> [original]
>>    F6 INVITE Example.com -> VM
>> 
>>    INVITE sip:vm@192.0.2.6;target=sip:carol%40example.com SIP/2.0
>>    Via: SIP/2.0/TCP proxy.example.com:5060;branch=z9hG4bKbbg4
>>    Via: SIP/2.0/TCP 192.0.2.3:5060;branch=z9hG4bK42t2
>>    Max-Forward: 67
>>    From: Alice <sip:alice@example.com>;tag=kkaz-
>>    To: Bob <sip:bob@example.com>
>>    Supported: histinfo
>>    Call-Id: 12345600@example.com
>>    CSeq: 1 INVITE
>>    History-Info: <sip:bob@example.com>;index=1
>>    History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302\
>>                       ;text="Moved Temporarily">;index=1.1;rc=1
>>    History-Info: <sip:carol@example.com?Reason=SIP%3Bcause%3D408>;\
>>                       index=1.2;mp=1
>>    History-Info: <sip:carol@192.0.2.4>;index=1.2.1;rc=1.2
>>    History-Info: <sip:vm@example.com;target=sip:carol%40example.com>;\
>>                       index=1.3;mp=1.2
>>    History-Info: <sip:vm@192.0.2.5;target=sip:carol%40example.com>\
>>                       ;cause=408;index=1.3.1;rc=1.3
>>    Contact: Alice <sip:alice@192.0.2.3>
>>    Content-Type: application/sdp
>>    Content-Length: <appropriate value>
>
>> [my indexing]
>>    INVITE sip:vm@192.0.2.6;target=sip:carol%40example.com;cause=408 SIP/2.0
>>    <snip/>
>>    History-Info: <sip:vm@example.com;target=sip:carol%40example.com;cause=408>;\
>>                       index=1.2.2;mp=1.2
>>    History-Info: <sip:vm@192.0.2.5;target=sip:carol%40example.com;cause=408>\
>>                       ;index=1.2.2.1;rc=1.2.2
>
>I believe that you've found an error.
>
>If the call was sent to *Bob's* voicemail, then the fork to VM would
>be a child of the INVITE to sip:bob@example.com (index 1), so the
>INVITE to voicemail would have index 1.3.
>
>But in this case, the call was sent to *Carol's* voicemail, it is
>(most likely) a child of the INVITE to sip:carol@example.com (index
>1.2), and so the INVITE to voicemail would have index 1.2.2.

I don't think the original indexing is an error.

Is a result of indexing unique?

In this case, the Contact of 302 has mp=1.2. The index "1.2" is
created by proxy.example.com. But if the index is "1.2.1.x" that created
by another proxy, what is the new index of the new INVITE?

1.3? 1.2.1.x.1??

And I have an additional question.
If Contact of 302 has no mp parameter, what is an index of the new hi-entry?
and what is a mp parameter of it.

Shinji.

From fandreas@cisco.com  Tue Feb 26 21:50:13 2013
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Subject: Re: [sipcore] Dispatching of draft-ivov-dispatch-trickle-ice-sip-00 and draft-ivov-dispatch-sdpfrag
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Ok - we'll add them to the MMUSIC agenda. Does any other group need to 
review these drafts at some point or can they be handled completely by 
MMUSIC ?

Thanks

-- Flemming

On 2/26/13 5:10 PM, Mary Barnes wrote:
> Hi all,
>
> We had our review of the DISPATCH-86 topics with the ADs this morning.
>   The decision was to dispatch these documents to the MMUSIC WG for
> discussion as potential work items:
> - draft-ivov-dispatch-trickle-ice-sip (currently proposes a new INFO
> event package)
> - draft-ivov-dispatch-sdpfrag (defines application/sdpfrag MIME media type)
>
> It made more sense to dispatch to MMUSIC since they are related to
> draft-ivov-mmusic-trickle-ice-00 which is currently under discussion
> as a proposed MMUSIC WG document.  And, neither the SIP change nor RFC
> 6086 require that this sort of work be done in SIPCORE.
>
> If you have any questions or comments, please let the ADs know ASAP.
> I will be posting this decision along with the plans for the other
> IETF-86 topics and agenda by tomorrow morning.
>
> Regards,
> Mary
> DISPATCH WG co-chair
> .
>


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On 2/26/13 11:50 PM, Flemming Andreasen wrote:
> Ok - we'll add them to the MMUSIC agenda. Does any other group need to 
> review these drafts at some point or can they be handled completely by 
> MMUSIC ?
That depends entirely on what the final mechanisms start to look like.

I would make sure SIPCORE stays aware of the things that end up 
transported in SIP, and as the drafts mature, continue to re-assess 
whether pieces of them need to be done in SIPCORE.
> Thanks
>
> -- Flemming
>
> On 2/26/13 5:10 PM, Mary Barnes wrote:
>> Hi all,
>>
>> We had our review of the DISPATCH-86 topics with the ADs this morning.
>>   The decision was to dispatch these documents to the MMUSIC WG for
>> discussion as potential work items:
>> - draft-ivov-dispatch-trickle-ice-sip (currently proposes a new INFO
>> event package)
>> - draft-ivov-dispatch-sdpfrag (defines application/sdpfrag MIME media 
>> type)
>>
>> It made more sense to dispatch to MMUSIC since they are related to
>> draft-ivov-mmusic-trickle-ice-00 which is currently under discussion
>> as a proposed MMUSIC WG document.  And, neither the SIP change nor RFC
>> 6086 require that this sort of work be done in SIPCORE.
>>
>> If you have any questions or comments, please let the ADs know ASAP.
>> I will be posting this decision along with the plans for the other
>> IETF-86 topics and agenda by tomorrow morning.
>>
>> Regards,
>> Mary
>> DISPATCH WG co-chair
>> .
>>
>


From worley@shell01.TheWorld.com  Wed Feb 27 08:28:58 2013
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From: worley@ariadne.com (Dale R. Worley)
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To: Flemming Andreasen <fandreas@cisco.com>
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Cc: fluffy@cisco.com, rai-ads@tools.ietf.org, rbarnes@bbn.com, sipcore@ietf.org, mmusic-chairs@tools.ietf.org
Subject: Re: [sipcore] Dispatching of draft-ivov-dispatch-trickle-ice-sip-00	and draft-ivov-dispatch-sdpfrag
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> From: Flemming Andreasen <fandreas@cisco.com>

> Ok - we'll add them to the MMUSIC agenda. Does any other group need to 
> review these drafts at some point or can they be handled completely by 
> MMUSIC ?

> From: Robert Sparks <rjsparks@nostrum.com>

> I would make sure SIPCORE stays aware of the things that end up 
> transported in SIP, and as the drafts mature, continue to re-assess 
> whether pieces of them need to be done in SIPCORE.

RTCWEB seems to want trickle ICE, and RTCWEB will be providing another
transport mechanism for trickle ICE updates, so I think they should
stay informed as well.

Dale

From adam@nostrum.com  Wed Feb 27 08:35:36 2013
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Cc: draft-rosen-rph-reg-policy@tools.ietf.org
Subject: [sipcore] Call for adoption/WGLC: draft-rosen-rph-reg-policy-00
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[as chair]

During the processing of draft-polk-local-emergency-rph-namespace, the 
IESG, authors, and WG chairs determined that the policy described in 
RFC4412, section 9 for defining a new namespace (Standards-Track RFC) 
was probably selected in error, and that "IETF Review" is likely more 
appropriate. The ECRIT working group has already been informed of this 
situation, and no parties have objected to a change in policy:

http://www.ietf.org/mail-archive/web/ecrit/current/msg08292.html

Since this is a policy change to a core SIP header, our Area Director 
has requested that the document that formally changes the policy be 
processed as a SIPCORE document. The current draft is available here:

http://tools.ietf.org/html/draft-rosen-rph-reg-policy-00

(Aside: ignoring the boilerplate, references, and table of contents, the 
document is about half as long as this email, and should take less time 
to read).

This message starts a call for adoption in parallel with a working group 
last call. Both periods end in just over two weeks, on Friday, March 
15th (the final Friday of the Orlando IETF meeting). Please reply to the 
mailing list indicating support or opposition to such adoption, and with 
any comments on the document contents themselves.

/a

From keith.drage@alcatel-lucent.com  Wed Feb 27 11:48:22 2013
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Adam Roach <adam@nostrum.com>, "'SIPCORE'" <sipcore@ietf.org>
Date: Wed, 27 Feb 2013 20:48:15 +0100
Thread-Topic: [sipcore] Call for adoption/WGLC: draft-rosen-rph-reg-policy-00
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Cc: "draft-rosen-rph-reg-policy@tools.ietf.org" <draft-rosen-rph-reg-policy@tools.ietf.org>
Subject: Re: [sipcore] Call for adoption/WGLC: draft-rosen-rph-reg-policy-00
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I do not have a problem with relaxing the criteria and therefore we will ne=
ed an I-D to do that, and therefore a milestone to cover that task.

I have the following comments on the document.

1)	Editorial. The document states internally that it updates RFC 4412 and t=
his is correct. However that information also needs to appear in the top le=
ft of the document.

2)	The document gives no background material on why IETF review has been ch=
osen, only that it no longer needs to be standards track (Note that I am no=
t convinced "Standards Track" was selected in error). Personally I think "S=
pecification Required" would be entirely appropriate, however if this is us=
ed see comment 3) below.

3)	As more relaxed criteria for IANA registration are defined, then materia=
l has to be defined to support those reviewing the registration; for a stan=
dards track defined protocol, this should be defined in a standards track d=
ocument. For example RFC 4412 contains no information on when a new namespa=
ce is appropriate, as opposed to reuse of an existing namespaces. I have ce=
rtainly been involved in discussions as to whether the "ETS" namespace is r=
estricted only to usage of "US government telecommunications service" or wh=
ether another government body in a different country with identical require=
ments can reuse the same namespace, or whether they have to define a new na=
mespace, and RFC 4412 does not answer this. So the proposed update currentl=
y contains no additional information (to RFC 4412) on:

-	what information needs to be provided for a new namespace or priority lev=
el;

-	what criteria need to be met for adopting a new namespace or priority lev=
el, versus reusing or modifying an existing one.

4)	Can a new document update an existing namespace with new priority levels=
 or other functionality? I believe this would normally be bad practice for =
compatibility reasons, but the document does not cover this.

Keith


> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behal=
f
> Of Adam Roach
> Sent: 27 February 2013 16:36
> To: 'SIPCORE'
> Cc: draft-rosen-rph-reg-policy@tools.ietf.org
> Subject: [sipcore] Call for adoption/WGLC: draft-rosen-rph-reg-policy-00
>=20
> [as chair]
>=20
> During the processing of draft-polk-local-emergency-rph-namespace, the
> IESG, authors, and WG chairs determined that the policy described in
> RFC4412, section 9 for defining a new namespace (Standards-Track RFC)
> was probably selected in error, and that "IETF Review" is likely more
> appropriate. The ECRIT working group has already been informed of this
> situation, and no parties have objected to a change in policy:
>=20
> http://www.ietf.org/mail-archive/web/ecrit/current/msg08292.html
>=20
> Since this is a policy change to a core SIP header, our Area Director
> has requested that the document that formally changes the policy be
> processed as a SIPCORE document. The current draft is available here:
>=20
> http://tools.ietf.org/html/draft-rosen-rph-reg-policy-00
>=20
> (Aside: ignoring the boilerplate, references, and table of contents, the
> document is about half as long as this email, and should take less time
> to read).
>=20
> This message starts a call for adoption in parallel with a working group
> last call. Both periods end in just over two weeks, on Friday, March
> 15th (the final Friday of the Orlando IETF meeting). Please reply to the
> mailing list indicating support or opposition to such adoption, and with
> any comments on the document contents themselves.
>=20
> /a
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore

From brian.rosen@neustar.biz  Wed Feb 27 12:13:36 2013
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From: "Rosen, Brian" <Brian.Rosen@neustar.biz>
To: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
Date: Wed, 27 Feb 2013 15:13:23 -0500
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Cc: "draft-rosen-rph-reg-policy@tools.ietf.org" <draft-rosen-rph-reg-policy@tools.ietf.org>, SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Call for adoption/WGLC: draft-rosen-rph-reg-policy-00
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We can fix 1 at AUTH48 if WGLC proceeds.

The document doesn't have to say why a particular management policy was cho=
sen.  It has to be the appropriate policy that the IETF agrees on.  In this=
 case, we think an Informational RFC is adequate to add a new namespace but=
 doesn't define new protocol behavior.  We think IETF Review is appropriate=
, so that we get review by the rather distributed constituency that cares a=
bout this particular piece of stuff.  I don't think Specification Required =
is adequate -- I want IETF experts to review.  I don't even think Expert Re=
view is adequate, because we do have several different classes of people wh=
o care about various aspects of this.

While I somewhat agree that 4412 is light on text defining when a new names=
pace is appropriate, it isn't the purpose of this document to rewrite that =
text, but rather to fix a specific problem that holds up other RFCs.  If th=
e policy is left at Standards Track, it still doesn't define when a new nam=
espace is appropriate.  It still doesn't define whether an update to an exi=
sting namespace is required.

We could start an effort to do a 4412-bis to address these issues, but I wo=
uld prefer not to turn this document into that.

Brian


On Feb 27, 2013, at 2:48 PM, "DRAGE, Keith (Keith)" <keith.drage@alcatel-lu=
cent.com> wrote:

> I do not have a problem with relaxing the criteria and therefore we will =
need an I-D to do that, and therefore a milestone to cover that task.
>=20
> I have the following comments on the document.
>=20
> 1)	Editorial. The document states internally that it updates RFC 4412 and=
 this is correct. However that information also needs to appear in the top =
left of the document.
>=20
> 2)	The document gives no background material on why IETF review has been =
chosen, only that it no longer needs to be standards track (Note that I am =
not convinced "Standards Track" was selected in error). Personally I think =
"Specification Required" would be entirely appropriate, however if this is =
used see comment 3) below.
>=20
> 3)	As more relaxed criteria for IANA registration are defined, then mater=
ial has to be defined to support those reviewing the registration; for a st=
andards track defined protocol, this should be defined in a standards track=
 document. For example RFC 4412 contains no information on when a new names=
pace is appropriate, as opposed to reuse of an existing namespaces. I have =
certainly been involved in discussions as to whether the "ETS" namespace is=
 restricted only to usage of "US government telecommunications service" or =
whether another government body in a different country with identical requi=
rements can reuse the same namespace, or whether they have to define a new =
namespace, and RFC 4412 does not answer this. So the proposed update curren=
tly contains no additional information (to RFC 4412) on:
>=20
> -	what information needs to be provided for a new namespace or priority l=
evel;
>=20
> -	what criteria need to be met for adopting a new namespace or priority l=
evel, versus reusing or modifying an existing one.
>=20
> 4)	Can a new document update an existing namespace with new priority leve=
ls or other functionality? I believe this would normally be bad practice fo=
r compatibility reasons, but the document does not cover this.
>=20
> Keith
>=20
>=20
>> -----Original Message-----
>> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Beha=
lf
>> Of Adam Roach
>> Sent: 27 February 2013 16:36
>> To: 'SIPCORE'
>> Cc: draft-rosen-rph-reg-policy@tools.ietf.org
>> Subject: [sipcore] Call for adoption/WGLC: draft-rosen-rph-reg-policy-00
>>=20
>> [as chair]
>>=20
>> During the processing of draft-polk-local-emergency-rph-namespace, the
>> IESG, authors, and WG chairs determined that the policy described in
>> RFC4412, section 9 for defining a new namespace (Standards-Track RFC)
>> was probably selected in error, and that "IETF Review" is likely more
>> appropriate. The ECRIT working group has already been informed of this
>> situation, and no parties have objected to a change in policy:
>>=20
>> http://www.ietf.org/mail-archive/web/ecrit/current/msg08292.html
>>=20
>> Since this is a policy change to a core SIP header, our Area Director
>> has requested that the document that formally changes the policy be
>> processed as a SIPCORE document. The current draft is available here:
>>=20
>> http://tools.ietf.org/html/draft-rosen-rph-reg-policy-00
>>=20
>> (Aside: ignoring the boilerplate, references, and table of contents, the
>> document is about half as long as this email, and should take less time
>> to read).
>>=20
>> This message starts a call for adoption in parallel with a working group
>> last call. Both periods end in just over two weeks, on Friday, March
>> 15th (the final Friday of the Orlando IETF meeting). Please reply to the
>> mailing list indicating support or opposition to such adoption, and with
>> any comments on the document contents themselves.
>>=20
>> /a
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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Subject: Re: [sipcore] Call for adoption/WGLC: draft-rosen-rph-reg-policy-00
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On 2/27/13 2:13 PM, Rosen, Brian wrote:
> We can fix 1 at AUTH48 if WGLC proceeds.
We'll have opportunity to revise the draft earlier than AUTH48.
>
> The document doesn't have to say why a particular management policy was chosen.  It has to be the appropriate policy that the IETF agrees on.  In this case, we think an Informational RFC is adequate to add a new namespace but doesn't define new protocol behavior.  We think IETF Review is appropriate, so that we get review by the rather distributed constituency that cares about this particular piece of stuff.  I don't think Specification Required is adequate -- I want IETF experts to review.  I don't even think Expert Review is adequate, because we do have several different classes of people who care about various aspects of this.
>
> While I somewhat agree that 4412 is light on text defining when a new namespace is appropriate, it isn't the purpose of this document to rewrite that text, but rather to fix a specific problem that holds up other RFCs.  If the policy is left at Standards Track, it still doesn't define when a new namespace is appropriate.  It still doesn't define whether an update to an existing namespace is required.
>
> We could start an effort to do a 4412-bis to address these issues, but I would prefer not to turn this document into that.
>
> Brian
>
>
> On Feb 27, 2013, at 2:48 PM, "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com> wrote:
>
>> I do not have a problem with relaxing the criteria and therefore we will need an I-D to do that, and therefore a milestone to cover that task.
>>
>> I have the following comments on the document.
>>
>> 1)	Editorial. The document states internally that it updates RFC 4412 and this is correct. However that information also needs to appear in the top left of the document.
>>
>> 2)	The document gives no background material on why IETF review has been chosen, only that it no longer needs to be standards track (Note that I am not convinced "Standards Track" was selected in error). Personally I think "Specification Required" would be entirely appropriate, however if this is used see comment 3) below.
IETF review is the better path in this case. We're still learning what 
the guidance we would give an expert reviewer (which would
go with specification required) - IETF review lets us move forward and 
discuss that as we go. If we actually do get many more
of these, and it becomes clear both what that guidance would look like 
and that we don't need IETF-wide review for other reasons, we could 
shift to Specification Required later.
>>
>> 3)	As more relaxed criteria for IANA registration are defined, then material has to be defined to support those reviewing the registration; for a standards track defined protocol, this should be defined in a standards track document. For example RFC 4412 contains no information on when a new namespace is appropriate, as opposed to reuse of an existing namespaces. I have certainly been involved in discussions as to whether the "ETS" namespace is restricted only to usage of "US government telecommunications service" or whether another government body in a different country with identical requirements can reuse the same namespace, or whether they have to define a new namespace, and RFC 4412 does not answer this. So the proposed update currently contains no additional information (to RFC 4412) on:
>>
>> -	what information needs to be provided for a new namespace or priority level;
>>
>> -	what criteria need to be met for adopting a new namespace or priority level, versus reusing or modifying an existing one.
>>
>> 4)	Can a new document update an existing namespace with new priority levels or other functionality? I believe this would normally be bad practice for compatibility reasons, but the document does not cover this.
>>
>> Keith
>>
>>
>>> -----Original Message-----
>>> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf
>>> Of Adam Roach
>>> Sent: 27 February 2013 16:36
>>> To: 'SIPCORE'
>>> Cc: draft-rosen-rph-reg-policy@tools.ietf.org
>>> Subject: [sipcore] Call for adoption/WGLC: draft-rosen-rph-reg-policy-00
>>>
>>> [as chair]
>>>
>>> During the processing of draft-polk-local-emergency-rph-namespace, the
>>> IESG, authors, and WG chairs determined that the policy described in
>>> RFC4412, section 9 for defining a new namespace (Standards-Track RFC)
>>> was probably selected in error, and that "IETF Review" is likely more
>>> appropriate. The ECRIT working group has already been informed of this
>>> situation, and no parties have objected to a change in policy:
>>>
>>> http://www.ietf.org/mail-archive/web/ecrit/current/msg08292.html
>>>
>>> Since this is a policy change to a core SIP header, our Area Director
>>> has requested that the document that formally changes the policy be
>>> processed as a SIPCORE document. The current draft is available here:
>>>
>>> http://tools.ietf.org/html/draft-rosen-rph-reg-policy-00
>>>
>>> (Aside: ignoring the boilerplate, references, and table of contents, the
>>> document is about half as long as this email, and should take less time
>>> to read).
>>>
>>> This message starts a call for adoption in parallel with a working group
>>> last call. Both periods end in just over two weeks, on Friday, March
>>> 15th (the final Friday of the Orlando IETF meeting). Please reply to the
>>> mailing list indicating support or opposition to such adoption, and with
>>> any comments on the document contents themselves.
>>>
>>> /a
>>> _______________________________________________
>>> sipcore mailing list
>>> sipcore@ietf.org
>>> https://www.ietf.org/mailman/listinfo/sipcore
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
> _______________________________________________
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From pkyzivat@alum.mit.edu  Wed Feb 27 14:16:11 2013
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Subject: Re: [sipcore] Call for adoption/WGLC: draft-rosen-rph-reg-policy-00
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Mostly WFM.

At the risk of setting off another long debate with Keith, ISTM that 
this document should be marked as updating 4412. (In the same way that 
draft-ietf-sipcore-priority updates 3261.) Otherwise, someone reading 
4412 will have no hint that the policies established in that document 
have been changed.

	Thanks,
	Paul

On 2/28/13 12:35 AM, Adam Roach wrote:
> [as chair]
>
> During the processing of draft-polk-local-emergency-rph-namespace, the
> IESG, authors, and WG chairs determined that the policy described in
> RFC4412, section 9 for defining a new namespace (Standards-Track RFC)
> was probably selected in error, and that "IETF Review" is likely more
> appropriate. The ECRIT working group has already been informed of this
> situation, and no parties have objected to a change in policy:
>
> http://www.ietf.org/mail-archive/web/ecrit/current/msg08292.html
>
> Since this is a policy change to a core SIP header, our Area Director
> has requested that the document that formally changes the policy be
> processed as a SIPCORE document. The current draft is available here:
>
> http://tools.ietf.org/html/draft-rosen-rph-reg-policy-00
>
> (Aside: ignoring the boilerplate, references, and table of contents, the
> document is about half as long as this email, and should take less time
> to read).
>
> This message starts a call for adoption in parallel with a working group
> last call. Both periods end in just over two weeks, on Friday, March
> 15th (the final Friday of the Orlando IETF meeting). Please reply to the
> mailing list indicating support or opposition to such adoption, and with
> any comments on the document contents themselves.
>
> /a
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>


From brian.rosen@neustar.biz  Wed Feb 27 14:18:04 2013
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From: "Rosen, Brian" <Brian.Rosen@neustar.biz>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
Date: Wed, 27 Feb 2013 17:17:56 -0500
Thread-Topic: [sipcore] Call for adoption/WGLC: draft-rosen-rph-reg-policy-00
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Subject: Re: [sipcore] Call for adoption/WGLC: draft-rosen-rph-reg-policy-00
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Yes, we will do that.

Brian

On Feb 27, 2013, at 5:13 PM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:

> Mostly WFM.
>=20
> At the risk of setting off another long debate with Keith, ISTM that this=
 document should be marked as updating 4412. (In the same way that draft-ie=
tf-sipcore-priority updates 3261.) Otherwise, someone reading 4412 will hav=
e no hint that the policies established in that document have been changed.
>=20
> 	Thanks,
> 	Paul
>=20
> On 2/28/13 12:35 AM, Adam Roach wrote:
>> [as chair]
>>=20
>> During the processing of draft-polk-local-emergency-rph-namespace, the
>> IESG, authors, and WG chairs determined that the policy described in
>> RFC4412, section 9 for defining a new namespace (Standards-Track RFC)
>> was probably selected in error, and that "IETF Review" is likely more
>> appropriate. The ECRIT working group has already been informed of this
>> situation, and no parties have objected to a change in policy:
>>=20
>> http://www.ietf.org/mail-archive/web/ecrit/current/msg08292.html
>>=20
>> Since this is a policy change to a core SIP header, our Area Director
>> has requested that the document that formally changes the policy be
>> processed as a SIPCORE document. The current draft is available here:
>>=20
>> http://tools.ietf.org/html/draft-rosen-rph-reg-policy-00
>>=20
>> (Aside: ignoring the boilerplate, references, and table of contents, the
>> document is about half as long as this email, and should take less time
>> to read).
>>=20
>> This message starts a call for adoption in parallel with a working group
>> last call. Both periods end in just over two weeks, on Friday, March
>> 15th (the final Friday of the Orlando IETF meeting). Please reply to the
>> mailing list indicating support or opposition to such adoption, and with
>> any comments on the document contents themselves.
>>=20
>> /a
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>=20
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From keith.drage@alcatel-lucent.com  Wed Feb 27 17:19:51 2013
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To: "Rosen, Brian" <Brian.Rosen@neustar.biz>, SIPCORE <sipcore@ietf.org>, "draft-rosen-rph-reg-policy@tools.ietf.org" <draft-rosen-rph-reg-policy@tools.ietf.org>
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Thread-Topic: [sipcore] Call for adoption/WGLC: draft-rosen-rph-reg-policy-00
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Subject: Re: [sipcore] Call for adoption/WGLC: draft-rosen-rph-reg-policy-00
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Brian wrote:
> I don't
> think Specification Required is adequate -- I want IETF experts to review=
.

If it is specification required then IETF experts will review.

      Specification Required - Values and their meanings must be
            documented in a permanent and readily available public
            specification, in sufficient detail so that interoperability
            between independent implementations is possible.  When used,
            Specification Required also implies use of a Designated
            Expert, who will review the public specification and
            evaluate whether it is sufficiently clear to allow
            interoperable implementations.  The intention behind
            "permanent and readily available" is that a document can
            reasonably be expected to be findable and retrievable long
            after IANA assignment of the requested value.  Publication
            of an RFC is an ideal means of achieving this requirement,
            but Specification Required is intended to also cover the
            case of a document published outside of the RFC path.  For
            RFC publication, the normal RFC review process is expected
            to provide the necessary review for interoperability, though
            the Designated Expert may be a particularly well-qualified
            person to perform such a review.

So to be clear, it gets review as follows:

-	by the experts involved in the original public specification by some othe=
r organization
-	by the designated expert

I am not sure that the designated expert has to be a single expert - it cou=
ld be several people - the ADs just need to put it in place.

Regards

Keith

> -----Original Message-----
> From: Rosen, Brian [mailto:Brian.Rosen@neustar.biz]
> Sent: 27 February 2013 20:13
> To: DRAGE, Keith (Keith)
> Cc: Adam Roach; SIPCORE; draft-rosen-rph-reg-policy@tools.ietf.org
> Subject: Re: [sipcore] Call for adoption/WGLC: draft-rosen-rph-reg-policy=
-
> 00
>=20
> We can fix 1 at AUTH48 if WGLC proceeds.
>=20
> The document doesn't have to say why a particular management policy was
> chosen.  It has to be the appropriate policy that the IETF agrees on.  In
> this case, we think an Informational RFC is adequate to add a new
> namespace but doesn't define new protocol behavior.  We think IETF Review
> is appropriate, so that we get review by the rather distributed
> constituency that cares about this particular piece of stuff.  I don't
> think Specification Required is adequate -- I want IETF experts to review=
.
> I don't even think Expert Review is adequate, because we do have several
> different classes of people who care about various aspects of this.
>=20
> While I somewhat agree that 4412 is light on text defining when a new
> namespace is appropriate, it isn't the purpose of this document to rewrit=
e
> that text, but rather to fix a specific problem that holds up other RFCs.
> If the policy is left at Standards Track, it still doesn't define when a
> new namespace is appropriate.  It still doesn't define whether an update
> to an existing namespace is required.
>=20
> We could start an effort to do a 4412-bis to address these issues, but I
> would prefer not to turn this document into that.
>=20
> Brian
>=20
>=20
> On Feb 27, 2013, at 2:48 PM, "DRAGE, Keith (Keith)" <keith.drage@alcatel-
> lucent.com> wrote:
>=20
> > I do not have a problem with relaxing the criteria and therefore we wil=
l
> need an I-D to do that, and therefore a milestone to cover that task.
> >
> > I have the following comments on the document.
> >
> > 1)	Editorial. The document states internally that it updates RFC 4412
> and this is correct. However that information also needs to appear in the
> top left of the document.
> >
> > 2)	The document gives no background material on why IETF review has
> been chosen, only that it no longer needs to be standards track (Note tha=
t
> I am not convinced "Standards Track" was selected in error). Personally I
> think "Specification Required" would be entirely appropriate, however if
> this is used see comment 3) below.
> >
> > 3)	As more relaxed criteria for IANA registration are defined, then
> material has to be defined to support those reviewing the registration;
> for a standards track defined protocol, this should be defined in a
> standards track document. For example RFC 4412 contains no information on
> when a new namespace is appropriate, as opposed to reuse of an existing
> namespaces. I have certainly been involved in discussions as to whether
> the "ETS" namespace is restricted only to usage of "US government
> telecommunications service" or whether another government body in a
> different country with identical requirements can reuse the same namespac=
e,
> or whether they have to define a new namespace, and RFC 4412 does not
> answer this. So the proposed update currently contains no additional
> information (to RFC 4412) on:
> >
> > -	what information needs to be provided for a new namespace or
> priority level;
> >
> > -	what criteria need to be met for adopting a new namespace or
> priority level, versus reusing or modifying an existing one.
> >
> > 4)	Can a new document update an existing namespace with new priority
> levels or other functionality? I believe this would normally be bad
> practice for compatibility reasons, but the document does not cover this.
> >
> > Keith
> >
> >
> >> -----Original Message-----
> >> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf
> >> Of Adam Roach
> >> Sent: 27 February 2013 16:36
> >> To: 'SIPCORE'
> >> Cc: draft-rosen-rph-reg-policy@tools.ietf.org
> >> Subject: [sipcore] Call for adoption/WGLC: draft-rosen-rph-reg-policy-
> 00
> >>
> >> [as chair]
> >>
> >> During the processing of draft-polk-local-emergency-rph-namespace, the
> >> IESG, authors, and WG chairs determined that the policy described in
> >> RFC4412, section 9 for defining a new namespace (Standards-Track RFC)
> >> was probably selected in error, and that "IETF Review" is likely more
> >> appropriate. The ECRIT working group has already been informed of this
> >> situation, and no parties have objected to a change in policy:
> >>
> >> http://www.ietf.org/mail-archive/web/ecrit/current/msg08292.html
> >>
> >> Since this is a policy change to a core SIP header, our Area Director
> >> has requested that the document that formally changes the policy be
> >> processed as a SIPCORE document. The current draft is available here:
> >>
> >> http://tools.ietf.org/html/draft-rosen-rph-reg-policy-00
> >>
> >> (Aside: ignoring the boilerplate, references, and table of contents,
> the
> >> document is about half as long as this email, and should take less tim=
e
> >> to read).
> >>
> >> This message starts a call for adoption in parallel with a working
> group
> >> last call. Both periods end in just over two weeks, on Friday, March
> >> 15th (the final Friday of the Orlando IETF meeting). Please reply to
> the
> >> mailing list indicating support or opposition to such adoption, and
> with
> >> any comments on the document contents themselves.
> >>
> >> /a
> >> _______________________________________________
> >> sipcore mailing list
> >> sipcore@ietf.org
> >> https://www.ietf.org/mailman/listinfo/sipcore
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore

