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Date: Fri, 1 Mar 2013 16:19:00 -0500 (EST)
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From: worley@ariadne.com (Dale R. Worley)
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In-reply-to: <580BEA5E3B99744AB1F5BFF5E9A3C67D16E36595DB@HE111648.emea1.cds.t-internal.com> (R.Jesske@telekom.de)
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Subject: Re: [sipcore] Verify draft-ietf-sipcore-rfc4244bis-callflows-02.txt
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I've gone through draft-ietf-sipcore-rfc4244bis-callflows-02.  I
haven't examine everything in the SIP messages.  Instead, I've
concentrated on the History-Info headers in the "final" messages of
the sequences.  In each example, there is a final message that
contains all of the information in all of the History-Info headers.  I
believe that the following amendments should be made in examles 3.1,
3.6, and 3.7.

These amendments include a number of things that have been discussed
on the Sipcore mailing list.

3.1.  Sequentially Forking (History-Info in Response)

   Alice   example.com            Bob     Office    Home
   |            |                  |        |        |
   | INVITE F1  |                  |        |        |
   |----------->|    INVITE F2     |        |        |
   |            |----------------->|        |        |
   | 100 Trying F3                 |        |        |
   |<-----------|  302 Move Temporarily F4  |        |
   |            |<-----------------|        |        |
   |            |   ACK F5         |        |        |
   |            |----------------->|        |        |
   |            |       INVITE F6           |        |
   |            |-------------------------->|        |
   |            |      180 Ringing F7       |        |
   |            |<--------------------------|        |
   |  180 Ringing F8                        |        |
   |<-----------|   retransmit INVITE       |        |
   |            |-------------------------->|        |
   |            |      ( timeout )          |        |
   |            |             INVITE F9              |
   |            |----------------------------------->|
   |            |           100 Trying F10           |
   |            |<-----------------------------------|
   |            |           486 Busy Here F11        |
   |            |<-----------------------------------|
   |  486 Busy Here F12                              |
   |<-----------|             ACK F13                |
   |            |----------------------------------->|
   |  ACK F14   |                                    |
   |----------->|                                    |

In these examples, I have added Reason values to hi-entries that have
effectively received failure responses due to the failure of their
internally redirected targets.  Doing so is optional per draft-4244bis
but it seems to me to be the best practice.

That is, I believe that the results of internal retargeting should be
the same as if the second retargeting was done in a different proxy.

   F12 486 Busy Here example.com -> alice

   History-Info: <sip:bob@example.com>;index=1
   History-Info: <sip:bob@192.0.2.4?Reason=SIP%3Bcause%3D302>;\
                 index=1.1;rc=1
-  History-Info: <sip:office@example.com>;index=1.2;mp=1
+  History-Info: <sip:office@example.com?Reason=SIP%3Bcause%3D408>;\
+                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
+                index=1.2.1;rc=1.2
   History-Info: <sip:home@example.com>;index=1.3;mp=1
   History-Info: <sip:home@192.0.2.6>;index=1.3.1;rc=1.3

3.6.  PBX Voicemail Example

  Alice      example.com       Bob          Carol        VM

  | INVITE F1    |              |             |          |
  |------------->|              |             |          |
  |              | INVITE  F2   |             |          |
  |              |------------->|             |          |
  |              |              |             |          |
  |  100 Trying  |              |             |          |
  |<-------------| 302 Moved Temporarily F3   |          |
  |              |<-------------|             |          |
  |              |              |             |          |
  |             ACK             |             |          |
  |---------------------------->|             |          |
  |              |              |             |          |
  |              | INVITE F4    |             |          |
  |              |--------------------------->|          |
  |              |              |             |          |
  |              |         180 Ringing  F5    |          |
  |              |<---------------------------|          |
  |              |              |             |          |
  | 180 Ringing  |              |             |          |
  |<-------------|              |             |          |
  |              |              |             |          |
  |              |       (timeout)            |          |
  |              |              |             |          |
  |              | INVITE  F6   |             |          |
  |              |-------------------------------------->|
  |              |              |             |          |
  |              |               200 OK  F7              |
  |              |<--------------------------------------|
  |   200 OK     |              |             |          |
  |<-------------|              |             |          |
  |              |              |             |          |
  |                         ACK                          |
  |----------------------------------------------------->|

Here, I assume we are using the "cause" URI parameter to indicate
redirection reasons per RFC 4458.  The redirection from Bob to Carol
is "Deflection immediate response", which is translated into the SIP
cause 480.  The redirection from Carol to VM is "No reply", which is
translated into SIP cause 408.  We assume the "cause" parameter is
preserved when AORs are mapped to their contacts.

Here, we choose the "cause" that is sent to VM to be 408, because the
immediately preceeding INVITE failed due to 408.  But the VM box is
Bob's, and the call to Bob failed due to cause=480.  The choice
probably depends on the application logic rather than standards.
(History-Info, however, captures the entire history of the call.)

The Reason value for index 1.2.1 has been corrected to 408.

  F7 200 OK VM -> Example.com

  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
+ History-Info: <sip:carol@example.com;cause=480?Reason=SIP%3Bcause%3D408>;\
+                    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

3.7.  Consumer Voicemail Example

   Alice      example.com       Bob          Carol        VM

   | INVITE F1    |              |             |          |
   |------------->|              |             |          |
   |              | INVITE  F2   |             |          |
   |              |------------->|             |          |
   |              |              |             |          |
   |  100 Trying  |              |             |          |
   |<-------------| 302 Moved Temporarily F3   |          |
   |              |<-------------|             |          |
   |              |              |             |          |
   |             ACK             |             |          |
   |---------------------------->|             |          |
   |              |              |             |          |
   |              | INVITE F4    |             |          |
   |              |--------------------------->|          |
   |              |              |             |          |
   |              |         180 Ringing  F5    |          |
   |              |<---------------------------|          |
   |              |              |             |          |
   | 180 Ringing  |              |             |          |
   |<-------------|              |             |          |
   |              |              |             |          |
   |              |       (timeout)            |          |
   |              |              |             |          |
   |              | INVITE  F6   |             |          |
   |              |-------------------------------------->|
   |              |              |             |          |
   |              |               200 OK  F7              |
   |              |<--------------------------------------|
   |   200 OK     |              |             |          |
   |<-------------|              |             |          |
   |              |              |             |          |
   |                         ACK                          |
   |----------------------------------------------------->|

Here, hi-entry 1.2.1 should have a Reason value, but the Reason value
on 1.2 is optional.

Since <sip:vm@example.com;target=sip:carol...> is generated from
<sip:carol@example.com> (index 1.2), it should have index 1.2.2, not
1.3.

   F7 200 OK VM -> Example.com

   History-Info: <sip:bob@example.com>;index=1
   History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302\
-                                                          ;text="Moved Temporarily">\
+                %3Btext%3D%22Moved%20Temporarily%22>\
                 ;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:carol@192.0.2.4?Reason=SIP%3Bcause%3D408>;\
+                index=1.2.1;rc=1.2
   History-Info: <sip:vm@example.com;target=sip:carol%40example.com>;\
-                      index=1.3;mp=1.2
+                      index=1.2.2;mp=1.2
   History-Info: <sip:vm@192.0.2.5;target=sip:carol%40example.com>;\
-                      index=1.3.1;rc=1.3
+                      index=1.2.2.1;rc=1.2.1

Dale

From wwwrun@rfc-editor.org  Fri Mar  1 16:01:57 2013
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Subject: [sipcore] RFC 6878 on IANA Registry for the Session Initiation Protocol (SIP) "Priority" Header Field
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A new Request for Comments is now available in online RFC libraries.

        
        RFC 6878

        Title:      IANA Registry for the Session 
                    Initiation Protocol (SIP) "Priority" Header Field 
        Author:     A.B. Roach
        Status:     Standards Track
        Stream:     IETF
        Date:       March 2013
        Mailbox:    adam@nostrum.com
        Pages:      3
        Characters: 4688
        Updates:    RFC3261

        I-D Tag:    draft-ietf-sipcore-priority-00.txt

        URL:        http://www.rfc-editor.org/rfc/rfc6878.txt

This document defines a new IANA registry to keep track of the values
defined for the Session Initiation Protocol (SIP) "Priority" header
field.  It updates RFC 3261.

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

This is now a Proposed Standard.

STANDARDS TRACK: This document specifies an Internet standards track
protocol for the Internet community,and requests discussion and suggestions
for improvements.  Please refer to the current edition of the Internet
Official Protocol Standards (STD 1) for the standardization state and
status of this protocol.  Distribution of this memo is unlimited.

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From rjsparks@nostrum.com  Mon Mar  4 14:05:38 2013
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Subject: Re: [sipcore] Call for adoption/WGLC: draft-rosen-rph-reg-policy-00
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repeating a thought just to make sure it was seen in this context

On 2/27/13 7:05 PM, DRAGE, Keith (Keith) wrote:
> 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 other organization
> -	by the designated expert
>
> I am not sure that the designated expert has to be a single expert - it could be several people - the ADs just need to put it in place.
That means we have a small number of people that we could identify to 
delegate review to for the whole community, and the community would 
trust they would see all the potential issues a proposal brings. In this 
case, I think we're not there yet, and
actually _do_ want full community review. As I said before, I think the 
right thing to do is take incremental steps with this one.
Get some experience with IETF review, and shift to something that only 
required expert review later.
>
> 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
>>
>> We can fix 1 at AUTH48 if WGLC proceeds.
>>
>> 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.
>>> 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
> _______________________________________________
> 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: Mon, 4 Mar 2013 23:21:23 +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, please read below:


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 websockets supposed to do if it =
gets a message containing Route header fields (think preloaded routes) wher=
e the next route value is for a transport=3Dws hop that a connection doesn'=
t yet exist for?

I replied to this subject in a previous mail, let me write it again (I
will extend it a bit more now):

Typical WebSocket capable SIP proxies/servers won't be able to
establish outgoing WS connections and, ever less, typical SIP WS
clients (browsers and probably mobile apps) 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, and the same for a UA that receives a 302
or a Refer-To pointing to a SIP URI with ;transport=3Dws.

Which is the best SIP response code for that kind of "error"? Well,
IMHO it does not exist so some existing proxies reply a custom 478
"Unsupported Transport" error when the request requires being routed
via an unsupported transport.

However I expect that SIP WebSocket clients should/will implement
Outbound and GRUU and will 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.




If this is ok I have the revision 08 ready to be submitted (which
includes the new "SIP Transport Registry" as requested by the WG).

Thanks a lot.


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

From rjsparks@nostrum.com  Mon Mar  4 14:32:47 2013
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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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On 3/4/13 4:21 PM, Iñaki Baz Castillo wrote:
> Hi Robert, please read below:
>
>
> 2013/2/25 Robert Sparks <rjsparks@nostrum.com>:
>> 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?
> I replied to this subject in a previous mail, let me write it again (I
> will extend it a bit more now):
Thanks, and sorry I couldn't find your earlier response.
>
> Typical WebSocket capable SIP proxies/servers won't be able to
> establish outgoing WS connections and, ever less, typical SIP WS
> clients (browsers and probably mobile apps) 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, and the same for a UA that receives a 302
> or a Refer-To pointing to a SIP URI with ;transport=ws.
>
> Which is the best SIP response code for that kind of "error"? Well,
> IMHO it does not exist so some existing proxies reply a custom 478
> "Unsupported Transport" error when the request requires being routed
> via an unsupported transport.
>
> However I expect that SIP WebSocket clients should/will implement
> Outbound and GRUU and will 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.
>
>
>
>
> If this is ok I have the revision 08 ready to be submitted (which
> includes the new "SIP Transport Registry" as requested by the WG).
Well, we've had some discussion here, but if I read what you're saying 
correctly,
you're not adding any discussion to the draft. I think there's value in 
doing so,
but we can keep talking about it during IETF LC.

Please submit the revision you have when submissions open Monday. Please 
be sure to address Keith's
concern: "The IANA considerations for the new registry needs to indicate 
it is grouped with the other SIP registries."
if you haven't already. What he's asking for is that we make an 
adjustment like this one:
OLD:
This document adds a new registry, "SIP Transport".
NEW:
This document adds a new registry, "SIP Transport" to the "Session 
Initiation Protocol (SIP) Parameters" registry page.

Thanks,

RjS
>
> Thanks a lot.
>
>


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<html>
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    <meta content="text/html; charset=UTF-8" http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    <div class="moz-cite-prefix">On 3/4/13 4:21 PM, Iñaki Baz Castillo
      wrote:<br>
    </div>
    <blockquote
cite="mid:CALiegfmbB01vEm-C5w2QXpcY86T4=+Qcdm-+epyJFgfKNoxPkA@mail.gmail.com"
      type="cite">
      <pre wrap="">Hi Robert, please read below:


2013/2/25 Robert Sparks <a class="moz-txt-link-rfc2396E" href="mailto:rjsparks@nostrum.com">&lt;rjsparks@nostrum.com&gt;</a>:
</pre>
      <blockquote type="cite">
        <pre wrap="">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:
</pre>
        <blockquote type="cite">
          <pre wrap="">
* 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?
</pre>
        </blockquote>
      </blockquote>
      <pre wrap="">
I replied to this subject in a previous mail, let me write it again (I
will extend it a bit more now):</pre>
    </blockquote>
    Thanks, and sorry I couldn't find your earlier response.<br>
    <blockquote
cite="mid:CALiegfmbB01vEm-C5w2QXpcY86T4=+Qcdm-+epyJFgfKNoxPkA@mail.gmail.com"
      type="cite">
      <pre wrap="">

Typical WebSocket capable SIP proxies/servers won't be able to
establish outgoing WS connections and, ever less, typical SIP WS
clients (browsers and probably mobile apps) 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, and the same for a UA that receives a 302
or a Refer-To pointing to a SIP URI with ;transport=ws.

Which is the best SIP response code for that kind of "error"? Well,
IMHO it does not exist so some existing proxies reply a custom 478
"Unsupported Transport" error when the request requires being routed
via an unsupported transport.

However I expect that SIP WebSocket clients should/will implement
Outbound and GRUU and will 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.




If this is ok I have the revision 08 ready to be submitted (which
includes the new "SIP Transport Registry" as requested by the WG).</pre>
    </blockquote>
    Well, we've had some discussion here, but if I read what you're
    saying correctly,<br>
    you're not adding any discussion to the draft. I think there's value
    in doing so,<br>
    but we can keep talking about it during IETF LC.<br>
    <br>
    Please submit the revision you have when submissions open Monday.
    Please be sure to address Keith's<br>
    concern: "<font color="navy" face="Arial" size="2"><span
        style="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."</span></font><br>
    if you haven't already. What he's asking for is that we make an
    adjustment like this one:<br>
    OLD:<br>
    This document adds a new registry, "SIP Transport".<br>
    NEW:<br>
    This document adds a new registry, "SIP Transport" to the "Session
    Initiation Protocol (SIP) Parameters" registry page.<br>
    <br>
    Thanks,<br>
    <br>
    RjS<br>
    <meta http-equiv="content-type" content="text/html; charset=UTF-8">
    <blockquote
cite="mid:CALiegfmbB01vEm-C5w2QXpcY86T4=+Qcdm-+epyJFgfKNoxPkA@mail.gmail.com"
      type="cite">
      <pre wrap="">

Thanks a lot.


</pre>
    </blockquote>
    <br>
  </body>
</html>

--------------000507090503060809090000--

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 5 Mar 2013 00:08:04 +0100
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Subject: Re: [sipcore] AD Review: draft-ietf-sipcore-sip-websocket-06.txt
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2013/3/4 Robert Sparks <rjsparks@nostrum.com>:
> Well, we've had some discussion here, but if I read what you're saying
> correctly,
> you're not adding any discussion to the draft. I think there's value in
> doing so,
> but we can keep talking about it during IETF LC.

I think that the same issue would occur in other cases as when a UA
establishes a dialog using SCTP as Contact URI transport. The remote
could use such a Contact into a Refer-To or 302 and same problem would
occur if the destination does not support SCTP transport.

Using an Outbound proxy (so hopefully RFC 5626) and GRUU mitigates
this problem. Even more, IMHO both Outbound and GRUU are the required
and standarized solution for all those scenarios which cover more than
just the new WS transport defined in the draft.



> Please submit the revision you have when submissions open Monday. Please =
be
> sure to address Keith's
> concern: "The IANA considerations for the new registry needs to indicate =
it
> is grouped with the other SIP registries."
> if you haven't already. What he's asking for is that we make an adjustmen=
t
> like this one:
> OLD:
>
> This document adds a new registry, "SIP Transport".
> NEW:
> This document adds a new registry, "SIP Transport" to the "Session
> Initiation Protocol (SIP) Parameters" registry page.

Thanks for pointing it.

So I will submit the new revision 08 on Monday.

Thanks a lot. Best regards.


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

From keith.drage@alcatel-lucent.com  Tue Mar  5 12:43:05 2013
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To: SIPCORE <sipcore@ietf.org>
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Thread-Topic: IPR Disclosure: SSH Communications Security Corporation's Statement	about IPR related to RFC 4028
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Subject: [sipcore] FW: IPR Disclosure: SSH Communications Security Corporation's Statement	about IPR related to RFC 4028
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Forwarding to the SIPCORE list, rather than condemning it to the old SIP li=
st.

Regards

Keith

-----Original Message-----
From: IETF Secretariat [mailto:ietf-ipr@ietf.org]=20
Sent: 05 March 2013 06:22
To: jdrosen@dynamicsoft.com; sdonovan@dynamicsoft.com
Cc: gonzalo.camarillo@ericsson.com; rjsparks@nostrum.com; DRAGE, Keith (Kei=
th); dean.willis@softarmor.com; sip@ietf.org; ipr-announce@ietf.org
Subject: IPR Disclosure: SSH Communications Security Corporation's Statemen=
t about IPR related to RFC 4028


Dear Jonathan Rosenberg, Steven Donovan:

 An IPR disclosure that pertains to your RFC entitled "Session Timers in th=
e
Session Initiation Protocol (SIP)" (RFC4028) was submitted to the IETF
Secretariat on 2013-02-23 and has been posted on the "IETF Page of Intellec=
tual
Property Rights Disclosures" (https://datatracker.ietf.org/ipr/1979/). The =
title
of the IPR disclosure is "SSH Communications Security Corporation's Stateme=
nt
about IPR related to RFC 4028."");

The IETF Secretariat


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Forwarding to the SIPCORE list, rather than condemning it to the old SIP li=
st.

Regards

Keith

-----Original Message-----
From: IETF Secretariat [mailto:ietf-ipr@ietf.org]=20
Sent: 05 March 2013 06:25
To: fluffy@cisco.com
Cc: gonzalo.camarillo@ericsson.com; rjsparks@nostrum.com; DRAGE, Keith (Kei=
th); dean.willis@softarmor.com; sip@ietf.org; ipr-announce@ietf.org
Subject: IPR Disclosure: SSH Communications Security Corporation's Statemen=
t about IPR related to RFC 5626


Dear Cullen Jennings:

 An IPR disclosure that pertains to your RFC entitled "Managing Client-Init=
iated
Connections in the Session Initiation Protocol (SIP)" (RFC5626) was submitt=
ed to
the IETF Secretariat on 2013-02-23 and has been posted on the "IETF Page of
Intellectual Property Rights Disclosures"
(https://datatracker.ietf.org/ipr/1992/). The title of the IPR disclosure i=
s
"SSH Communications Security Corporation's Statement about IPR related to R=
FC
5626."");

The IETF Secretariat


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From: Ofer Goren <oferg@avaya.com>
To: "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: Question on RFC 5407 (crossover of BYE and UPDATE)
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Hi

I checked RFC 5407 for race condition in various scenarios but I failed to =
find any indication of crossover between BYE and UPDATE.
So I have a connected callLeg, and I send an UPDATE transaction from A to B=
. At the same time, I send BYE from B to A.
Should A re-transmit the UPDATE in case B did not answer even after A repli=
ed with 200OK for the BYE?
What about other general transactions (such as OPTION)? Is UPDATE special i=
n this case and should be treated differently than other general transactio=
ns on that call?

Thanks,
Ofer


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<p class=3D"MsoNormal">Hi<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">I checked RFC 5407 for race condition in various sce=
narios but I failed to find any indication of crossover between BYE and UPD=
ATE.<o:p></o:p></p>
<p class=3D"MsoNormal">So I have a connected callLeg, and I send an UPDATE =
transaction from A to B. At the same time, I send BYE from B to A.<o:p></o:=
p></p>
<p class=3D"MsoNormal">Should A re-transmit the UPDATE in case B did not an=
swer even after A replied with 200OK for the BYE?<o:p></o:p></p>
<p class=3D"MsoNormal">What about other general transactions (such as OPTIO=
N)? Is UPDATE special in this case and should be treated differently than o=
ther general transactions on that call?<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>
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one point.

worley@ariadne.com (Dale R. Worley)
Fri, 1 Mar 2013 16:19:00 -0500 (EST)
<snip/>
>3.7.  Consumer Voicemail Example
<snip/>
>   History-Info: <sip:vm@192.0.2.5;target=sip:carol%40example.com>;\
>-                      index=1.3.1;rc=1.3
>+                      index=1.2.2.1;rc=1.2.1
                                         ^^^^^
                                         1.2.2
Shinji.
             

From christer.holmberg@ericsson.com  Thu Mar  7 03:26:43 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Ofer Goren <oferg@avaya.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Thread-Topic: Question on RFC 5407 (crossover of BYE and UPDATE)
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Subject: Re: [sipcore] Question on RFC 5407 (crossover of BYE and UPDATE)
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Hi,

As the BYE terminates the dialog, I see no reason why you should re-transit=
 the UPDATE, or any other mid-dialog requests associated with that dialog, =
as the dialog doesn't exist anymore.

Regards,

Christer

From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf =
Of Ofer Goren
Sent: 7. maaliskuuta 2013 11:20
To: sipcore@ietf.org
Subject: [sipcore] Question on RFC 5407 (crossover of BYE and UPDATE)

Hi

I checked RFC 5407 for race condition in various scenarios but I failed to =
find any indication of crossover between BYE and UPDATE.
So I have a connected callLeg, and I send an UPDATE transaction from A to B=
. At the same time, I send BYE from B to A.
Should A re-transmit the UPDATE in case B did not answer even after A repli=
ed with 200OK for the BYE?
What about other general transactions (such as OPTION)? Is UPDATE special i=
n this case and should be treated differently than other general transactio=
ns on that call?

Thanks,
Ofer


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<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Hi,<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">As the BYE terminates =
the dialog, I see no reason why you should re-transit the UPDATE, or any ot=
her mid-dialog requests associated with that dialog, as the dialog doesn&#8=
217;t exist anymore.<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">Regards,<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">Christer<o:p></o:p></s=
pan></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 0cm =
0cm 0cm">
<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> 7. maaliskuuta 2013 11:20<br>
<b>To:</b> sipcore@ietf.org<br>
<b>Subject:</b> [sipcore] Question on RFC 5407 (crossover of BYE and UPDATE=
)<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Hi<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">I checked RFC 5407 for race condition in various sce=
narios but I failed to find any indication of crossover between BYE and UPD=
ATE.<o:p></o:p></p>
<p class=3D"MsoNormal">So I have a connected callLeg, and I send an UPDATE =
transaction from A to B. At the same time, I send BYE from B to A.<o:p></o:=
p></p>
<p class=3D"MsoNormal">Should A re-transmit the UPDATE in case B did not an=
swer even after A replied with 200OK for the BYE?<o:p></o:p></p>
<p class=3D"MsoNormal">What about other general transactions (such as OPTIO=
N)? Is UPDATE special in this case and should be treated differently than o=
ther general transactions on that call?<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>
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From rjsparks@nostrum.com  Thu Mar  7 06:42:22 2013
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>, Ofer Goren <oferg@avaya.com>
Subject: Re: [sipcore] Question on RFC 5407 (crossover of BYE and UPDATE)
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On 3/7/13 5:26 AM, Christer Holmberg wrote:
>
> Hi,
>
> As the BYE terminates the dialog, I see no reason why you should 
> re-transit the UPDATE, or any other mid-dialog requests associated 
> with that dialog, as the dialog doesn't exist anymore.
>
Transactions complete independently. If you were to follow the 
specifications, you will retransmit the UPDATE until it gets a final 
response or the appropriate timer fires ending the transaction.

The real problem is that the request may actually have gotten to the UAS 
before the BYE was processed, but the response to it is what got lost. 
Your retransmission will stimulate retransmission of that response, and 
you need to receive that response in order to have a consistent view of 
state at both endpoints.

Christer - it's not hard to imagine an applicaitons use of an INFO 
package that you would break with the optimization you propose.

RjS
>
> Regards,
>
> Christer
>
> *From:*sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] *On 
> Behalf Of *Ofer Goren
> *Sent:* 7. maaliskuuta 2013 11:20
> *To:* sipcore@ietf.org
> *Subject:* [sipcore] Question on RFC 5407 (crossover of BYE and UPDATE)
>
> Hi
>
> I checked RFC 5407 for race condition in various scenarios but I 
> failed to find any indication of crossover between BYE and UPDATE.
>
> So I have a connected callLeg, and I send an UPDATE transaction from A 
> to B. At the same time, I send BYE from B to A.
>
> Should A re-transmit the UPDATE in case B did not answer even after A 
> replied with 200OK for the BYE?
>
> What about other general transactions (such as OPTION)? Is UPDATE 
> special in this case and should be treated differently than other 
> general transactions on that call?
>
> Thanks,
>
> Ofer
>
>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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    <div class="moz-cite-prefix">On 3/7/13 5:26 AM, Christer Holmberg
      wrote:<br>
    </div>
    <blockquote
      cite="mid:7594FB04B1934943A5C02806D1A2204B11496B@ESESSMB209.ericsson.se"
      type="cite">
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      <div class="WordSection1">
        <p class="MsoNormal"><span style="color:#1F497D">Hi,<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D">As the BYE
            terminates the dialog, I see no reason why you should
            re-transit the UPDATE, or any other mid-dialog requests
            associated with that dialog, as the dialog doesn&#8217;t exist
            anymore.</span></p>
      </div>
    </blockquote>
    Transactions complete independently. If you were to follow the
    specifications, you will retransmit the UPDATE until it gets a final
    response or the appropriate timer fires ending the transaction.<br>
    <br>
    The real problem is that the request may actually have gotten to the
    UAS before the BYE was processed, but the response to it is what got
    lost. Your retransmission will stimulate retransmission of that
    response, and you need to receive that response in order to have a
    consistent view of state at both endpoints.<br>
    <br>
    Christer - it's not hard to imagine an applicaitons use of an INFO
    package that you would break with the optimization you propose.<br>
    <br>
    RjS<br>
    <blockquote
      cite="mid:7594FB04B1934943A5C02806D1A2204B11496B@ESESSMB209.ericsson.se"
      type="cite">
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        <p class="MsoNormal"><span style="color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D">Regards,<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D">Christer<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
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          <div style="border:none;border-top:solid #B5C4DF
            1.0pt;padding:3.0pt 0cm 0cm 0cm">
            <p class="MsoNormal"><b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">
                <a class="moz-txt-link-abbreviated" href="mailto:sipcore-bounces@ietf.org">sipcore-bounces@ietf.org</a>
                [<a class="moz-txt-link-freetext" href="mailto:sipcore-bounces@ietf.org">mailto:sipcore-bounces@ietf.org</a>]
                <b>On Behalf Of </b>Ofer Goren<br>
                <b>Sent:</b> 7. maaliskuuta 2013 11:20<br>
                <b>To:</b> <a class="moz-txt-link-abbreviated" href="mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
                <b>Subject:</b> [sipcore] Question on RFC 5407
                (crossover of BYE and UPDATE)<o:p></o:p></span></p>
          </div>
        </div>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <p class="MsoNormal">Hi<o:p></o:p></p>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <p class="MsoNormal">I checked RFC 5407 for race condition in
          various scenarios but I failed to find any indication of
          crossover between BYE and UPDATE.<o:p></o:p></p>
        <p class="MsoNormal">So I have a connected callLeg, and I send
          an UPDATE transaction from A to B. At the same time, I send
          BYE from B to A.<o:p></o:p></p>
        <p class="MsoNormal">Should A re-transmit the UPDATE in case B
          did not answer even after A replied with 200OK for the BYE?<o:p></o:p></p>
        <p class="MsoNormal">What about other general transactions (such
          as OPTION)? Is UPDATE special in this case and should be
          treated differently than other general transactions on that
          call?<o:p></o:p></p>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <p class="MsoNormal">Thanks,<o:p></o:p></p>
        <p class="MsoNormal">Ofer<o:p></o:p></p>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
      </div>
      <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>
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From christer.holmberg@ericsson.com  Thu Mar  7 06:52:42 2013
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Robert Sparks <rjsparks@nostrum.com>
Thread-Topic: [sipcore] Question on RFC 5407 (crossover of BYE and UPDATE)
Thread-Index: Ac4bFCZJoEunDd07Rd+FdHFYonFvOQAEjI9QAATPKYAAAlzVYA==
Date: Thu, 7 Mar 2013 14:52:35 +0000
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References: <D9AD3D1627F52341B829F32E6DDE089149E25848@rvil-mail2010> <7594FB04B1934943A5C02806D1A2204B11496B@ESESSMB209.ericsson.se> <5138A749.80202@nostrum.com>
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>, Ofer Goren <oferg@avaya.com>
Subject: Re: [sipcore] Question on RFC 5407 (crossover of BYE and UPDATE)
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Hi,

My point was more that any state synchronization etc shouldn't matter, as t=
he dialog is anyway terminated.

But, Robert is correct, if you follow the specification (lots of implementa=
tions don't :), and re-transmitting until you get a response will make sure=
 you're on the "safe side".

Regards,

Christer

From: Robert Sparks [mailto:rjsparks@nostrum.com]
Sent: 7. maaliskuuta 2013 16:42
To: Christer Holmberg
Cc: Ofer Goren; sipcore@ietf.org
Subject: Re: [sipcore] Question on RFC 5407 (crossover of BYE and UPDATE)

On 3/7/13 5:26 AM, Christer Holmberg wrote:
Hi,

As the BYE terminates the dialog, I see no reason why you should re-transit=
 the UPDATE, or any other mid-dialog requests associated with that dialog, =
as the dialog doesn't exist anymore.
Transactions complete independently. If you were to follow the specificatio=
ns, you will retransmit the UPDATE until it gets a final response or the ap=
propriate timer fires ending the transaction.

The real problem is that the request may actually have gotten to the UAS be=
fore the BYE was processed, but the response to it is what got lost. Your r=
etransmission will stimulate retransmission of that response, and you need =
to receive that response in order to have a consistent view of state at bot=
h endpoints.

Christer - it's not hard to imagine an applicaitons use of an INFO package =
that you would break with the optimization you propose.

RjS


Regards,

Christer

From: sipcore-bounces@ietf.org<mailto:sipcore-bounces@ietf.org> [mailto:sip=
core-bounces@ietf.org] On Behalf Of Ofer Goren
Sent: 7. maaliskuuta 2013 11:20
To: sipcore@ietf.org<mailto:sipcore@ietf.org>
Subject: [sipcore] Question on RFC 5407 (crossover of BYE and UPDATE)

Hi

I checked RFC 5407 for race condition in various scenarios but I failed to =
find any indication of crossover between BYE and UPDATE.
So I have a connected callLeg, and I send an UPDATE transaction from A to B=
. At the same time, I send BYE from B to A.
Should A re-transmit the UPDATE in case B did not answer even after A repli=
ed with 200OK for the BYE?
What about other general transactions (such as OPTION)? Is UPDATE special i=
n this case and should be treated differently than other general transactio=
ns on that call?

Thanks,
Ofer





_______________________________________________

sipcore mailing list

sipcore@ietf.org<mailto:sipcore@ietf.org>

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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Hi,<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">My point was more that=
 any state synchronization etc shouldn&#8217;t matter, as the dialog is any=
way terminated.<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">But, Robert is correct=
, if you follow the specification (lots of implementations don&#8217;t :), =
and re-transmitting until you get a response will make sure you&#8217;re on=
 the &#8220;safe side&#8221;.<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">Regards,<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">Christer<o:p></o:p></s=
pan></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 0cm =
0cm 0cm">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">From:</span></b><spa=
n style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif=
&quot;;color:windowtext"> Robert Sparks [mailto:rjsparks@nostrum.com]
<br>
<b>Sent:</b> 7. maaliskuuta 2013 16:42<br>
<b>To:</b> Christer Holmberg<br>
<b>Cc:</b> Ofer Goren; sipcore@ietf.org<br>
<b>Subject:</b> Re: [sipcore] Question on RFC 5407 (crossover of BYE and UP=
DATE)<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On 3/7/13 5:26 AM, Christer Holmberg wrote:<o:p></o:=
p></p>
</div>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Hi,</span><o:p></o:p><=
/p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&nbsp;</span><o:p></o:=
p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">As the BYE terminates =
the dialog, I see no reason why you should re-transit the UPDATE, or any ot=
her mid-dialog requests associated with that dialog, as the dialog doesn&#8=
217;t exist anymore.</span><o:p></o:p></p>
</blockquote>
<p class=3D"MsoNormal"><span style=3D"font-size:12.0pt;font-family:&quot;Ti=
mes New Roman&quot;,&quot;serif&quot;">Transactions complete independently.=
 If you were to follow the specifications, you will retransmit the UPDATE u=
ntil it gets a final response or the appropriate timer fires
 ending the transaction.<br>
<br>
The real problem is that the request may actually have gotten to the UAS be=
fore the BYE was processed, but the response to it is what got lost. Your r=
etransmission will stimulate retransmission of that response, and you need =
to receive that response in order
 to have a consistent view of state at both endpoints.<br>
<br>
Christer - it's not hard to imagine an applicaitons use of an INFO package =
that you would break with the optimization you propose.<br>
<br>
RjS<br>
<br>
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&nbsp;</span><o:p></o:=
p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Regards,</span><o:p></=
o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&nbsp;</span><o:p></o:=
p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Christer</span><o:p></=
o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&nbsp;</span><o:p></o:=
p></p>
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm">
<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> 7. maaliskuuta 2013 11:20<br>
<b>To:</b> <a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
<b>Subject:</b> [sipcore] Question on RFC 5407 (crossover of BYE and UPDATE=
)</span><o:p></o:p></p>
</div>
</div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal">Hi<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal">I checked RFC 5407 for race condition in various sce=
narios but I failed to find any indication of crossover between BYE and UPD=
ATE.<o:p></o:p></p>
<p class=3D"MsoNormal">So I have a connected callLeg, and I send an UPDATE =
transaction from A to B. At the same time, I send BYE from B to A.<o:p></o:=
p></p>
<p class=3D"MsoNormal">Should A re-transmit the UPDATE in case B did not an=
swer even after A replied with 200OK for the BYE?<o:p></o:p></p>
<p class=3D"MsoNormal">What about other general transactions (such as OPTIO=
N)? Is UPDATE special in this case and should be treated differently than o=
ther general transactions on that call?<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;<o:p></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">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal"><span style=3D"font-size:12.0pt;font-family:&quot;Ti=
mes New Roman&quot;,&quot;serif&quot;"><br>
<br>
<br>
<o:p></o:p></span></p>
<pre>_______________________________________________<o:p></o:p></pre>
<pre>sipcore mailing list<o:p></o:p></pre>
<pre><a href=3D"mailto:sipcore@ietf.org">sipcore@ietf.org</a><o:p></o:p></p=
re>
<pre><a href=3D"https://www.ietf.org/mailman/listinfo/sipcore">https://www.=
ietf.org/mailman/listinfo/sipcore</a><o:p></o:p></pre>
<p class=3D"MsoNormal"><span style=3D"font-size:12.0pt;font-family:&quot;Ti=
mes New Roman&quot;,&quot;serif&quot;"><o:p>&nbsp;</o:p></span></p>
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Date: Thu, 07 Mar 2013 10:02:56 -0600
From: Robert Sparks <rjsparks@nostrum.com>
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>, Ofer Goren <oferg@avaya.com>
Subject: Re: [sipcore] Question on RFC 5407 (crossover of BYE and UPDATE)
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On 3/7/13 8:52 AM, Christer Holmberg wrote:
>
> Hi,
>
> My point was more that any state synchronization etc shouldn't matter, 
> as the dialog is anyway terminated.
>
I think you are assuming that the state you are synchronizing is limited 
to dialog state.
There are application side-effects of processing many of these messages 
that you can't predict,
and the applications really do care about the result of that transaction 
at the other end.
>
> But, Robert is correct, if you follow the specification (lots of 
> implementations don't :),
>
And risk causing grief for doing so.
>
> and re-transmitting until you get a response will make sure you're on 
> the "safe side".
>
> Regards,
>
> Christer
>
> *From:*Robert Sparks [mailto:rjsparks@nostrum.com]
> *Sent:* 7. maaliskuuta 2013 16:42
> *To:* Christer Holmberg
> *Cc:* Ofer Goren; sipcore@ietf.org
> *Subject:* Re: [sipcore] Question on RFC 5407 (crossover of BYE and 
> UPDATE)
>
> On 3/7/13 5:26 AM, Christer Holmberg wrote:
>
>     Hi,
>
>     As the BYE terminates the dialog, I see no reason why you should
>     re-transit the UPDATE, or any other mid-dialog requests associated
>     with that dialog, as the dialog doesn't exist anymore.
>
> Transactions complete independently. If you were to follow the 
> specifications, you will retransmit the UPDATE until it gets a final 
> response or the appropriate timer fires ending the transaction.
>
> The real problem is that the request may actually have gotten to the 
> UAS before the BYE was processed, but the response to it is what got 
> lost. Your retransmission will stimulate retransmission of that 
> response, and you need to receive that response in order to have a 
> consistent view of state at both endpoints.
>
> Christer - it's not hard to imagine an applicaitons use of an INFO 
> package that you would break with the optimization you propose.
>
> RjS
>
> Regards,
>
> Christer
>
> *From:*sipcore-bounces@ietf.org <mailto:sipcore-bounces@ietf.org> 
> [mailto:sipcore-bounces@ietf.org] *On Behalf Of *Ofer Goren
> *Sent:* 7. maaliskuuta 2013 11:20
> *To:* sipcore@ietf.org <mailto:sipcore@ietf.org>
> *Subject:* [sipcore] Question on RFC 5407 (crossover of BYE and UPDATE)
>
> Hi
>
> I checked RFC 5407 for race condition in various scenarios but I 
> failed to find any indication of crossover between BYE and UPDATE.
>
> So I have a connected callLeg, and I send an UPDATE transaction from A 
> to B. At the same time, I send BYE from B to A.
>
> Should A re-transmit the UPDATE in case B did not answer even after A 
> replied with 200OK for the BYE?
>
> What about other general transactions (such as OPTION)? Is UPDATE 
> special in this case and should be treated differently than other 
> general transactions on that call?
>
> Thanks,
>
> Ofer
>
>
>
>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org  <mailto:sipcore@ietf.org>
> https://www.ietf.org/mailman/listinfo/sipcore
>


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    <div class="moz-cite-prefix">On 3/7/13 8:52 AM, Christer Holmberg
      wrote:<br>
    </div>
    <blockquote
      cite="mid:7594FB04B1934943A5C02806D1A2204B114D5F@ESESSMB209.ericsson.se"
      type="cite">
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      <div class="WordSection1">
        <p class="MsoNormal"><span style="color:#1F497D">Hi,<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D">My point was
            more that any state synchronization etc shouldn&#8217;t matter, as
            the dialog is anyway terminated.</span></p>
      </div>
    </blockquote>
    I think you are assuming that the state you are synchronizing is
    limited to dialog state.<br>
    There are application side-effects of processing many of these
    messages that you can't predict,<br>
    and the applications really do care about the result of that
    transaction at the other end.<br>
    <blockquote
      cite="mid:7594FB04B1934943A5C02806D1A2204B114D5F@ESESSMB209.ericsson.se"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"><span style="color:#1F497D"><o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D">But, Robert is
            correct, if you follow the specification (lots of
            implementations don&#8217;t :),</span></p>
      </div>
    </blockquote>
    And risk causing grief for doing so.<br>
    <blockquote
      cite="mid:7594FB04B1934943A5C02806D1A2204B114D5F@ESESSMB209.ericsson.se"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"><span style="color:#1F497D"> and
            re-transmitting until you get a response will make sure
            you&#8217;re on the &#8220;safe side&#8221;.<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D">Regards,<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D">Christer<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D"><o:p>&nbsp;</o:p></span></p>
        <div>
          <div style="border:none;border-top:solid #B5C4DF
            1.0pt;padding:3.0pt 0cm 0cm 0cm">
            <p class="MsoNormal"><b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">From:</span></b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">
                Robert Sparks [<a class="moz-txt-link-freetext" href="mailto:rjsparks@nostrum.com">mailto:rjsparks@nostrum.com</a>]
                <br>
                <b>Sent:</b> 7. maaliskuuta 2013 16:42<br>
                <b>To:</b> Christer Holmberg<br>
                <b>Cc:</b> Ofer Goren; <a class="moz-txt-link-abbreviated" href="mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
                <b>Subject:</b> Re: [sipcore] Question on RFC 5407
                (crossover of BYE and UPDATE)<o:p></o:p></span></p>
          </div>
        </div>
        <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        <div>
          <p class="MsoNormal">On 3/7/13 5:26 AM, Christer Holmberg
            wrote:<o:p></o:p></p>
        </div>
        <blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
          <p class="MsoNormal"><span style="color:#1F497D">Hi,</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="color:#1F497D">&nbsp;</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="color:#1F497D">As the BYE
              terminates the dialog, I see no reason why you should
              re-transit the UPDATE, or any other mid-dialog requests
              associated with that dialog, as the dialog doesn&#8217;t exist
              anymore.</span><o:p></o:p></p>
        </blockquote>
        <p class="MsoNormal"><span
            style="font-size:12.0pt;font-family:&quot;Times New
            Roman&quot;,&quot;serif&quot;">Transactions complete
            independently. If you were to follow the specifications, you
            will retransmit the UPDATE until it gets a final response or
            the appropriate timer fires ending the transaction.<br>
            <br>
            The real problem is that the request may actually have
            gotten to the UAS before the BYE was processed, but the
            response to it is what got lost. Your retransmission will
            stimulate retransmission of that response, and you need to
            receive that response in order to have a consistent view of
            state at both endpoints.<br>
            <br>
            Christer - it's not hard to imagine an applicaitons use of
            an INFO package that you would break with the optimization
            you propose.<br>
            <br>
            RjS<br>
            <br>
            <o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color:#1F497D">&nbsp;</span><o:p></o:p></p>
        <p class="MsoNormal"><span style="color:#1F497D">Regards,</span><o:p></o:p></p>
        <p class="MsoNormal"><span style="color:#1F497D">&nbsp;</span><o:p></o:p></p>
        <p class="MsoNormal"><span style="color:#1F497D">Christer</span><o:p></o:p></p>
        <p class="MsoNormal"><span style="color:#1F497D">&nbsp;</span><o:p></o:p></p>
        <div>
          <div style="border:none;border-top:solid #B5C4DF
            1.0pt;padding:3.0pt 0cm 0cm 0cm">
            <p class="MsoNormal"><b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">
                <a moz-do-not-send="true"
                  href="mailto:sipcore-bounces@ietf.org">sipcore-bounces@ietf.org</a>
                [<a moz-do-not-send="true"
                  href="mailto:sipcore-bounces@ietf.org">mailto:sipcore-bounces@ietf.org</a>]
                <b>On Behalf Of </b>Ofer Goren<br>
                <b>Sent:</b> 7. maaliskuuta 2013 11:20<br>
                <b>To:</b> <a moz-do-not-send="true"
                  href="mailto:sipcore@ietf.org">sipcore@ietf.org</a><br>
                <b>Subject:</b> [sipcore] Question on RFC 5407
                (crossover of BYE and UPDATE)</span><o:p></o:p></p>
          </div>
        </div>
        <p class="MsoNormal">&nbsp;<o:p></o:p></p>
        <p class="MsoNormal">Hi<o:p></o:p></p>
        <p class="MsoNormal">&nbsp;<o:p></o:p></p>
        <p class="MsoNormal">I checked RFC 5407 for race condition in
          various scenarios but I failed to find any indication of
          crossover between BYE and UPDATE.<o:p></o:p></p>
        <p class="MsoNormal">So I have a connected callLeg, and I send
          an UPDATE transaction from A to B. At the same time, I send
          BYE from B to A.<o:p></o:p></p>
        <p class="MsoNormal">Should A re-transmit the UPDATE in case B
          did not answer even after A replied with 200OK for the BYE?<o:p></o:p></p>
        <p class="MsoNormal">What about other general transactions (such
          as OPTION)? Is UPDATE special in this case and should be
          treated differently than other general transactions on that
          call?<o:p></o:p></p>
        <p class="MsoNormal">&nbsp;<o:p></o:p></p>
        <p class="MsoNormal">Thanks,<o:p></o:p></p>
        <p class="MsoNormal">Ofer<o:p></o:p></p>
        <p class="MsoNormal">&nbsp;<o:p></o:p></p>
        <p class="MsoNormal"><span
            style="font-size:12.0pt;font-family:&quot;Times New
            Roman&quot;,&quot;serif&quot;"><br>
            <br>
            <br>
            <o:p></o:p></span></p>
        <pre>_______________________________________________<o:p></o:p></pre>
        <pre>sipcore mailing list<o:p></o:p></pre>
        <pre><a moz-do-not-send="true" href="mailto:sipcore@ietf.org">sipcore@ietf.org</a><o:p></o:p></pre>
        <pre><a moz-do-not-send="true" href="https://www.ietf.org/mailman/listinfo/sipcore">https://www.ietf.org/mailman/listinfo/sipcore</a><o:p></o:p></pre>
        <p class="MsoNormal"><span
            style="font-size:12.0pt;font-family:&quot;Times New
            Roman&quot;,&quot;serif&quot;"><o:p>&nbsp;</o:p></span></p>
      </div>
    </blockquote>
    <br>
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From shida@ntt-at.com  Thu Mar  7 17:02:41 2013
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Subject: Re: [sipcore] Verify draft-ietf-sipcore-rfc4244bis-callflows-02.txt
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Thanks Dale and Shinji;

 I will reflect these in the upcoming update.=20

 Thanks
  Shida

On Mar 1, 2013, at 1:19 PM, Dale R. Worley wrote:

> I've gone through draft-ietf-sipcore-rfc4244bis-callflows-02.  I
> haven't examine everything in the SIP messages.  Instead, I've
> concentrated on the History-Info headers in the "final" messages of
> the sequences.  In each example, there is a final message that
> contains all of the information in all of the History-Info headers.  I
> believe that the following amendments should be made in examles 3.1,
> 3.6, and 3.7.
>=20
> These amendments include a number of things that have been discussed
> on the Sipcore mailing list.
>=20
> 3.1.  Sequentially Forking (History-Info in Response)
>=20
>   Alice   example.com            Bob     Office    Home
>   |            |                  |        |        |
>   | INVITE F1  |                  |        |        |
>   |----------->|    INVITE F2     |        |        |
>   |            |----------------->|        |        |
>   | 100 Trying F3                 |        |        |
>   |<-----------|  302 Move Temporarily F4  |        |
>   |            |<-----------------|        |        |
>   |            |   ACK F5         |        |        |
>   |            |----------------->|        |        |
>   |            |       INVITE F6           |        |
>   |            |-------------------------->|        |
>   |            |      180 Ringing F7       |        |
>   |            |<--------------------------|        |
>   |  180 Ringing F8                        |        |
>   |<-----------|   retransmit INVITE       |        |
>   |            |-------------------------->|        |
>   |            |      ( timeout )          |        |
>   |            |             INVITE F9              |
>   |            |----------------------------------->|
>   |            |           100 Trying F10           |
>   |            |<-----------------------------------|
>   |            |           486 Busy Here F11        |
>   |            |<-----------------------------------|
>   |  486 Busy Here F12                              |
>   |<-----------|             ACK F13                |
>   |            |----------------------------------->|
>   |  ACK F14   |                                    |
>   |----------->|                                    |
>=20
> In these examples, I have added Reason values to hi-entries that have
> effectively received failure responses due to the failure of their
> internally redirected targets.  Doing so is optional per draft-4244bis
> but it seems to me to be the best practice.
>=20
> That is, I believe that the results of internal retargeting should be
> the same as if the second retargeting was done in a different proxy.
>=20
>   F12 486 Busy Here example.com -> alice
>=20
>   History-Info: <sip:bob@example.com>;index=3D1
>   History-Info: <sip:bob@192.0.2.4?Reason=3DSIP%3Bcause%3D302>;\
>                 index=3D1.1;rc=3D1
> -  History-Info: <sip:office@example.com>;index=3D1.2;mp=3D1
> +  History-Info: <sip:office@example.com?Reason=3DSIP%3Bcause%3D408>;\
> +                index=3D1.2;mp=3D1
>   History-Info: <sip:office@192.0.2.5?Reason=3DSIP%3Bcause%3D408>;\
> -                index=3D1.2.1;index=3D1.2.1;rc=3D1.2
> +                index=3D1.2.1;rc=3D1.2
>   History-Info: <sip:home@example.com>;index=3D1.3;mp=3D1
>   History-Info: <sip:home@192.0.2.6>;index=3D1.3.1;rc=3D1.3
>=20
> 3.6.  PBX Voicemail Example
>=20
>  Alice      example.com       Bob          Carol        VM
>=20
>  | INVITE F1    |              |             |          |
>  |------------->|              |             |          |
>  |              | INVITE  F2   |             |          |
>  |              |------------->|             |          |
>  |              |              |             |          |
>  |  100 Trying  |              |             |          |
>  |<-------------| 302 Moved Temporarily F3   |          |
>  |              |<-------------|             |          |
>  |              |              |             |          |
>  |             ACK             |             |          |
>  |---------------------------->|             |          |
>  |              |              |             |          |
>  |              | INVITE F4    |             |          |
>  |              |--------------------------->|          |
>  |              |              |             |          |
>  |              |         180 Ringing  F5    |          |
>  |              |<---------------------------|          |
>  |              |              |             |          |
>  | 180 Ringing  |              |             |          |
>  |<-------------|              |             |          |
>  |              |              |             |          |
>  |              |       (timeout)            |          |
>  |              |              |             |          |
>  |              | INVITE  F6   |             |          |
>  |              |-------------------------------------->|
>  |              |              |             |          |
>  |              |               200 OK  F7              |
>  |              |<--------------------------------------|
>  |   200 OK     |              |             |          |
>  |<-------------|              |             |          |
>  |              |              |             |          |
>  |                         ACK                          |
>  |----------------------------------------------------->|
>=20
> Here, I assume we are using the "cause" URI parameter to indicate
> redirection reasons per RFC 4458.  The redirection from Bob to Carol
> is "Deflection immediate response", which is translated into the SIP
> cause 480.  The redirection from Carol to VM is "No reply", which is
> translated into SIP cause 408.  We assume the "cause" parameter is
> preserved when AORs are mapped to their contacts.
>=20
> Here, we choose the "cause" that is sent to VM to be 408, because the
> immediately preceeding INVITE failed due to 408.  But the VM box is
> Bob's, and the call to Bob failed due to cause=3D480.  The choice
> probably depends on the application logic rather than standards.
> (History-Info, however, captures the entire history of the call.)
>=20
> The Reason value for index 1.2.1 has been corrected to 408.
>=20
>  F7 200 OK VM -> Example.com
>=20
>  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?Reason=3DSIP%3Bcause%3D480>;\
> -                    index=3D1.2.1;rc=3D1.2
> - History-Info: <sip:vm@example.com;\
> -                    target=3Dsip:bob%40example.com>;\
> -                    index=3D1.3;mp=3D1
> - History-Info: <sip:vm@192.0.2.6;\
> -                    target=3Dsip:bob%40example.com>;\
> -                    index=3D1.3.1;rc=3D1.3
> + History-Info: =
<sip:carol@example.com;cause=3D480?Reason=3DSIP%3Bcause%3D408>;\
> +                    index=3D1.2;mp=3D1
> + History-Info: =
<sip:carol@192.0.2.4;cause=3D480?Reason=3DSIP%3Bcause%3D408>;\
> +                    index=3D1.2.1;rc=3D1.2
> + History-Info: <sip:vm@example.com;\
> +                    target=3Dsip:bob%40example.com;cause=3D408>;\
> +                    index=3D1.3;mp=3D1
> + History-Info: <sip:vm@192.0.2.6;\
> +                    target=3Dsip:bob%40example.com;cause=3D408>;\
> +                    index=3D1.3.1;rc=3D1.3
>=20
> 3.7.  Consumer Voicemail Example
>=20
>   Alice      example.com       Bob          Carol        VM
>=20
>   | INVITE F1    |              |             |          |
>   |------------->|              |             |          |
>   |              | INVITE  F2   |             |          |
>   |              |------------->|             |          |
>   |              |              |             |          |
>   |  100 Trying  |              |             |          |
>   |<-------------| 302 Moved Temporarily F3   |          |
>   |              |<-------------|             |          |
>   |              |              |             |          |
>   |             ACK             |             |          |
>   |---------------------------->|             |          |
>   |              |              |             |          |
>   |              | INVITE F4    |             |          |
>   |              |--------------------------->|          |
>   |              |              |             |          |
>   |              |         180 Ringing  F5    |          |
>   |              |<---------------------------|          |
>   |              |              |             |          |
>   | 180 Ringing  |              |             |          |
>   |<-------------|              |             |          |
>   |              |              |             |          |
>   |              |       (timeout)            |          |
>   |              |              |             |          |
>   |              | INVITE  F6   |             |          |
>   |              |-------------------------------------->|
>   |              |              |             |          |
>   |              |               200 OK  F7              |
>   |              |<--------------------------------------|
>   |   200 OK     |              |             |          |
>   |<-------------|              |             |          |
>   |              |              |             |          |
>   |                         ACK                          |
>   |----------------------------------------------------->|
>=20
> Here, hi-entry 1.2.1 should have a Reason value, but the Reason value
> on 1.2 is optional.
>=20
> Since <sip:vm@example.com;target=3Dsip:carol...> is generated from
> <sip:carol@example.com> (index 1.2), it should have index 1.2.2, not
> 1.3.
>=20
>   F7 200 OK VM -> Example.com
>=20
>   History-Info: <sip:bob@example.com>;index=3D1
>   History-Info: <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302\
> -                                                          =
;text=3D"Moved Temporarily">\
> +                %3Btext%3D%22Moved%20Temporarily%22>\
>                 ;index=3D1.1;rc=3D1
>   History-Info: <sip:carol@example.com?Reason=3DSIP%3Bcause%3D408>;\
>                      index=3D1.2;mp=3D1
> -  History-Info: <sip:carol@192.0.2.4>;index=3D1.2.1;rc=3D1.2
> +  History-Info: <sip:carol@192.0.2.4?Reason=3DSIP%3Bcause%3D408>;\
> +                index=3D1.2.1;rc=3D1.2
>   History-Info: <sip:vm@example.com;target=3Dsip:carol%40example.com>;\
> -                      index=3D1.3;mp=3D1.2
> +                      index=3D1.2.2;mp=3D1.2
>   History-Info: <sip:vm@192.0.2.5;target=3Dsip:carol%40example.com>;\
> -                      index=3D1.3.1;rc=3D1.3
> +                      index=3D1.2.2.1;rc=3D1.2.1
>=20
> Dale
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From shinji.okumura@softfront.jp  Thu Mar  7 23:16:10 2013
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From: OKUMURA Shinji <shinji.okumura@softfront.jp>
To: Christer Holmberg <christer.holmberg@ericsson.com>
Date: Fri, 08 Mar 2013 16:16:03 +0900
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>, Ofer Goren <oferg@avaya.com>
Subject: Re: [sipcore] Question on RFC 5407 (crossover of BYE and UPDATE)
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Hi,

Additionally, if any subscription exist, the dialog does not terminate.

Shinji.

Christer Holmberg <christer.holmberg@ericsson.com>
Thu, 7 Mar 2013 11:26:37 +0000
>Hi,
>
>As the BYE terminates the dialog, I see no reason why you should re-transit the UPDATE, or any other mid-dialog requests 
>associated with that dialog, as the dialog doesn't exist anymore.
>
>Regards,
>
>Christer
>
>
>From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf Of Ofer Goren
>Sent: 7. maaliskuuta 2013 11:20
>To: sipcore@ietf.org
>Subject: [sipcore] Question on RFC 5407 (crossover of BYE and UPDATE)
>
>
>Hi
>
>I checked RFC 5407 for race condition in various scenarios but I failed to find any indication of crossover between BYE and 
>UPDATE.
>So I have a connected callLeg, and I send an UPDATE transaction from A to B. At the same time, I send BYE from B to A.
>Should A re-transmit the UPDATE in case B did not answer even after A replied with 200OK for the BYE?
>What about other general transactions (such as OPTION)? Is UPDATE special in this case and should be treated differently than 
>other general transactions on that call?
>
>Thanks,
>Ofer

From oferg@avaya.com  Fri Mar  8 15:44:54 2013
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From: Ofer Goren <oferg@avaya.com>
To: OKUMURA Shinji <shinji.okumura@softfront.jp>
Thread-Topic: [sipcore] Question on RFC 5407 (crossover of BYE and UPDATE)
Thread-Index: Ac4bFCZJoEunDd07Rd+FdHFYonFvOQAEjI9QACVrr4AAJq74/g==
Date: Fri, 8 Mar 2013 23:43:40 +0000
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Subject: Re: [sipcore] Question on RFC 5407 (crossover of BYE and UPDATE)
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--_000_D885B128C85C4AC292F21D8A2352F2AEavayacom_
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Thank you all for the clear answers.

Sent from IPhone 7


=E1-8 =E1=EE=F8=F5 2013, =E1=F9=F2=E4 09:15, "OKUMURA Shinji" <shinji.okumu=
ra@softfront.jp<mailto:shinji.okumura@softfront.jp>> =EB=FA=E1/=E4:

Hi,

Additionally, if any subscription exist, the dialog does not terminate.

Shinji.

Christer Holmberg <christer.holmberg@ericsson.com<mailto:christer.holmberg@=
ericsson.com>>
Thu, 7 Mar 2013 11:26:37 +0000
Hi,

As the BYE terminates the dialog, I see no reason why you should re-transit=
 the UPDATE, or any other mid-dialog requests
associated with that dialog, as the dialog doesn't exist anymore.

Regards,

Christer


From: sipcore-bounces@ietf.org<mailto:sipcore-bounces@ietf.org> [mailto:sip=
core-bounces@ietf.org] On Behalf Of Ofer Goren
Sent: 7. maaliskuuta 2013 11:20
To: sipcore@ietf.org<mailto:sipcore@ietf.org>
Subject: [sipcore] Question on RFC 5407 (crossover of BYE and UPDATE)


Hi

I checked RFC 5407 for race condition in various scenarios but I failed to =
find any indication of crossover between BYE and
UPDATE.
So I have a connected callLeg, and I send an UPDATE transaction from A to B=
. At the same time, I send BYE from B to A.
Should A re-transmit the UPDATE in case B did not answer even after A repli=
ed with 200OK for the BYE?
What about other general transactions (such as OPTION)? Is UPDATE special i=
n this case and should be treated differently than
other general transactions on that call?

Thanks,
Ofer
_______________________________________________
sipcore mailing list
sipcore@ietf.org<mailto:sipcore@ietf.org>
https://www.ietf.org/mailman/listinfo/sipcore

--_000_D885B128C85C4AC292F21D8A2352F2AEavayacom_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Diso-8859-=
8-i">
</head>
<body dir=3D"auto">
<div>Thank you all for the clear answers.<br>
<br>
<div style=3D"text-align: left;direction: ltr; ">Sent from IPhone 7</div>
<div style=3D"text-align: left;direction: ltr; "><br>
</div>
</div>
<div><br>
=E1-8 =E1=EE=F8=F5 2013, =E1=F9=F2=E4 09:15, &quot;OKUMURA Shinji&quot; &lt=
;<a href=3D"mailto:shinji.okumura@softfront.jp">shinji.okumura@softfront.jp=
</a>&gt; =EB=FA=E1/=E4:<br>
<br>
</div>
<blockquote type=3D"cite">
<div><span>Hi,</span><br>
<span></span><br>
<span>Additionally, if any subscription exist, the dialog does not terminat=
e.</span><br>
<span></span><br>
<span>Shinji.</span><br>
<span></span><br>
<span>Christer Holmberg &lt;<a href=3D"mailto:christer.holmberg@ericsson.co=
m">christer.holmberg@ericsson.com</a>&gt;</span><br>
<span>Thu, 7 Mar 2013 11:26:37 &#43;0000</span><br>
<blockquote type=3D"cite"><span>Hi,</span><br>
</blockquote>
<blockquote type=3D"cite"><span></span><br>
</blockquote>
<blockquote type=3D"cite"><span>As the BYE terminates the dialog, I see no =
reason why you should re-transit the UPDATE, or any other mid-dialog reques=
ts
</span><br>
</blockquote>
<blockquote type=3D"cite"><span>associated with that dialog, as the dialog =
doesn't exist anymore.</span><br>
</blockquote>
<blockquote type=3D"cite"><span></span><br>
</blockquote>
<blockquote type=3D"cite"><span>Regards,</span><br>
</blockquote>
<blockquote type=3D"cite"><span></span><br>
</blockquote>
<blockquote type=3D"cite"><span>Christer</span><br>
</blockquote>
<blockquote type=3D"cite"><span></span><br>
</blockquote>
<blockquote type=3D"cite"><span></span><br>
</blockquote>
<blockquote type=3D"cite"><span>From: <a href=3D"mailto:sipcore-bounces@iet=
f.org">sipcore-bounces@ietf.org</a> [<a href=3D"mailto:sipcore-bounces@ietf=
.org">mailto:sipcore-bounces@ietf.org</a>] On Behalf Of Ofer Goren</span><b=
r>
</blockquote>
<blockquote type=3D"cite"><span>Sent: 7. maaliskuuta 2013 11:20</span><br>
</blockquote>
<blockquote type=3D"cite"><span>To: <a href=3D"mailto:sipcore@ietf.org">sip=
core@ietf.org</a></span><br>
</blockquote>
<blockquote type=3D"cite"><span>Subject: [sipcore] Question on RFC 5407 (cr=
ossover of BYE and UPDATE)</span><br>
</blockquote>
<blockquote type=3D"cite"><span></span><br>
</blockquote>
<blockquote type=3D"cite"><span></span><br>
</blockquote>
<blockquote type=3D"cite"><span>Hi</span><br>
</blockquote>
<blockquote type=3D"cite"><span></span><br>
</blockquote>
<blockquote type=3D"cite"><span>I checked RFC 5407 for race condition in va=
rious scenarios but I failed to find any indication of crossover between BY=
E and
</span><br>
</blockquote>
<blockquote type=3D"cite"><span>UPDATE.</span><br>
</blockquote>
<blockquote type=3D"cite"><span>So I have a connected callLeg, and I send a=
n UPDATE transaction from A to B. At the same time, I send BYE from B to A.=
</span><br>
</blockquote>
<blockquote type=3D"cite"><span>Should A re-transmit the UPDATE in case B d=
id not answer even after A replied with 200OK for the BYE?</span><br>
</blockquote>
<blockquote type=3D"cite"><span>What about other general transactions (such=
 as OPTION)? Is UPDATE special in this case and should be treated different=
ly than
</span><br>
</blockquote>
<blockquote type=3D"cite"><span>other general transactions on that call?</s=
pan><br>
</blockquote>
<blockquote type=3D"cite"><span></span><br>
</blockquote>
<blockquote type=3D"cite"><span>Thanks,</span><br>
</blockquote>
<blockquote type=3D"cite"><span>Ofer</span><br>
</blockquote>
<span>_______________________________________________</span><br>
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Subject: [sipcore] I-D Action: draft-ietf-sipcore-sip-websocket-08.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-08.txt
	Pages           : 22
	Date            : 2013-03-11

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-08

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-sipcore-sip-websocket-08


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


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 11 Mar 2013 11:40:00 +0100
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2013/3/11  <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-08.txt
>         Pages           : 22
>         Date            : 2013-03-11
>
> 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-08
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-sipcore-sip-websocket-08



Hi all. The changelog in this new revision is the following:

- Added a "SIP Transport Registry" under "IANA Considerations" section
(thanks to Robert Sparks and Paul Kyzivat).


Best regards.






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

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--NextPart

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

    Title         : Session Initiation Protocol (SIP) History-Info Header Call Flow Examples
    Author(s)     : M. Barnes, et al
    Filename      : draft-ietf-sipcore-rfc4244bis-callflows
    Pages         : 46 
    Date          : March 19, 2013 
    
   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.


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From worley@shell01.TheWorld.com  Thu Mar 21 15:03:24 2013
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From: worley@ariadne.com (Dale R. Worley)
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To: OKUMURA Shinji <shinji.okumura@softfront.jp>
In-reply-to: <29CE1B24928B3Ashinji.okumura@softfront.jp>
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Cc: sipcore@ietf.org
Subject: Re: [sipcore] Verify draft-ietf-sipcore-rfc4244bis-callflows-02.txt
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> From: OKUMURA Shinji <shinji.okumura@softfront.jp>
> Date: Thu, 07 Mar 2013 20:11:53 +0900
> 
> one point.
> 
> worley@ariadne.com (Dale R. Worley)
> Fri, 1 Mar 2013 16:19:00 -0500 (EST)
> <snip/>
> >3.7.  Consumer Voicemail Example
> <snip/>
> >   History-Info: <sip:vm@192.0.2.5;target=sip:carol%40example.com>;\
> >-                      index=1.3.1;rc=1.3
> >+                      index=1.2.2.1;rc=1.2.1
>                                          ^^^^^
>                                          1.2.2

Yes, you are right about that.

Dale

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From: worley@ariadne.com (Dale R. Worley)
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To: sipcore@ietf.org
Subject: [sipcore] Copy edits for draft-ietf-sipcore-rfc4244bis-callflows-03
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This is a bunch of copy-edits to
draft-ietf-sipcore-rfc4244bis-callflows-03.  Section pointers are
marked with "*" at the left margin.  Individual items are marked with
"." at the left margin.

(Finding the inconsistencies between the various History-Info values
was made infinitely easier by a Perl script, "hi-consistency", which
is appended.  If you give that script the -03 draft, the output
flags inconsistent values of the H-I headers within each section of
the draft.  The output it generates from -03 is also appended.)

* All of the figure headings have the "figure N" part doubled:

            Figure 1: Figure 1: Example with Sequential Forking
          Figure 2: Figure 2: Example with Privacy Header Fields
    Figure 3: Figure 3: Example with Privacy Header Field for Specific
        Figure 4: Figure 4: Example for Automatic Call Distribution
                     Figure 5: Figure 5: Alias Example
             Figure 6: Figure 6: Enterprise Voivemail Example
              Figure 7: Figure 7: Consumer Voivemail Example
                     Figure 8: Figure 8: GRUU Example
              Figure 9: Figure 9: Limited Use Address Example
               Figure 10: Figure 10: Service Number Example

(I expect the RFC Editor can correct this.)

* Section 3.1

. From F8, I see that the rule is to record 1xx responses as well as
final failure responses:

       History-Info: <sip:office@192.0.2.5;Reason=SIP%3Bcause%3D180>;\
                                                         index=1.2.1;rc=1.2

(If recording 1xx responses with Reason is not intended, please modify
the following items by removing any Reason header-value that specifies
a 1xx cause value.)

. Correct the punctuation:

       F8 180 Ringing example.com -> alice

    -  History-Info: <sip:office@192.0.2.5;Reason=SIP%3Bcause%3D180>;\
    +  History-Info: <sip:office@192.0.2.5?Reason=SIP%3Bcause%3D180>;\
                                                         index=1.2.1;rc=1.2

. Remove duplicate index parameter:

       F9 INVITE example.com -> home

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

. The handling of the Reason on index=1.2 seems to be inconsistent:

Branch 1.2 is first seen in INVITE F6.  (INVITE F6 itself is
index=1.2.1, generated by internal forking from branch 1.2.)

   F6 INVITE example.com -> office

   History-Info: <sip:office@example.com>;index=1.2;mp=1

INVITE F9 is branch 1.3.1, generated by internal forking from branch
1.3, which was generated due to the failure of INVITE F6 (by the
timeout between F8 and F9).  In F9, index=1.2.1 is shown with a Reason
value (which is mandatory), but index=1.2 is not (which is allowed,
but not best practice).

   F9 INVITE example.com -> home

   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

INVITE F11 follows the pattern of F9.

   F11 486 Busy Here home -> example.com

   History-Info: <sip:office@example.com>;index=1.2;mp=1

But response F12 shows the Reason value on index=1.2 that is best
practice.

   F12 486 Busy Here example.com -> alice

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

I think what is intended is for F8, F9 and F11 to have the Reason
value on
index=1.2 as well.

       F8 180 Ringing example.com -> alice

    -  History-Info: <sip:office@example.com>;index=1.2;mp=1
    +  History-Info:
    <sip:office@example.com?Reason=SIP%3Bcause%3D180>;\
    +                                                  index=1.2;mp=1

       F9 INVITE example.com -> home

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

       F11 486 Busy Here home -> example.com

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

* Section 3.4

. Make F5 consistent with preceding messages:

       F5 INVITE Silver.Example.com -> Agent

       History-Info:
       <sip:Gold@gold.example.com?Reason=SIP%3Bcause%3D302>;\
    -                  index=1.1
    +                  rc=1;index=1.1

Same for F6, F7, and F8.

* Section 3.6

. URIs for 1.2 and 1.2.1 are not consistent:

   F4 INVITE Example.com -> Carol

   History-Info: <sip:carol@example.com>;index=1.2;mp=1
   History-Info: <sip:carol@192.0.2.4>;index=1.2.1;rc=1.2

   F5 180 Ringing Carol -> Example.com

   History-Info: <sip:carol@example.com>;index=1.2;mp=1
   History-Info: <sip:carol@192.0.2.4>;index=1.2.1;rc=1.2

   F6 INVITE Example.com -> VM

   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

   F7 200 OK VM -> Example.com

   History-Info:
   <sip:carol@example.com;cause=480?Reason=SIP%3Bcause%3D\
                      408>;index=1.2;mp=1
   History-Info: <sip:carol@192.0.2.4;cause=480?Reason=SIP%3Bcause%3D\
                      408>;index=1.2.1;rc=1.2

My understanding is that the intention is to show use of the cause URI
parameter (not the cause value in Reason!) per RFC 4458.  (See
Marianne Mohali's explanation of the situation at
http://www.ietf.org/mail-archive/web/sipcore/current/msg05514.html.)
If that is so, the URI for branch 1.2 would be generated as shown in
F7, <sip:carol@example.com;cause=480>, and presumably the parameter
would be carried into the URI for branch 1.2.1.  That requires the
changes shown below.

. Also, in INVITE F6, the timeout (408) response to branches 1.2
and 1.2.1 would already be recorded, as they are in response F7 (which
is generated from INVITE F6!).  Those changes are also included below.

. Also, in response F7, the cause URI parameter recorded in branches
1.3 and 1.3.1 is not the same as in INVITE F6:  F6 has "cause=480" and
F7 has "cause=408".  Which code to choose depends on the application
logic (see the discussion around Marianne Mohali's message referenced
above), and I have no particular opinion one way or the other, but the
two messages must be consistent.  No change for that is included
below.

       F4 INVITE Example.com -> Carol

    -  History-Info: <sip:carol@example.com>;index=1.2;mp=1
    -  History-Info: <sip:carol@192.0.2.4>;index=1.2.1;rc=1.2
    +  History-Info: <sip:carol@example.com;cause=480>;index=1.2;mp=1
    +  History-Info:
    -  <sip:carol@192.0.2.4;cause=480>;index=1.2.1;rc=1.2

       F5 180 Ringing Carol -> Example.com

    -  History-Info: <sip:carol@example.com>;index=1.2;mp=1
    -  History-Info: <sip:carol@192.0.2.4>;index=1.2.1;rc=1.2
    +  History-Info: <sip:carol@example.com;cause=480>;index=1.2;mp=1
    +  History-Info:
    -  <sip:carol@192.0.2.4;cause=480>;index=1.2.1;rc=1.2

       F6 INVITE Example.com -> VM

    -  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:carol@example.com;cause=480?Reason=SIP%3Bcause%3D\
    +                     408>;index=1.2;mp=1
    +  History-Info:
    -  <sip:carol@192.0.2.4;cause=480?Reason=SIP%3Bcause%3D\
    +                     408>;index=1.2.1;rc=1.2

       F7 200 OK VM -> Example.com

       History-Info:
       <sip:carol@example.com;cause=480?Reason=SIP%3Bcause%3D\
                          408>;index=1.2;mp=1
       History-Info:
       <sip:carol@192.0.2.4;cause=480?Reason=SIP%3Bcause%3D\
                          408>;index=1.2.1;rc=1.2

* Section 3.7

. There are a number of H-I entries that want to encode a cause text
"Moved Temporarily".  More exactly, they encode a URI header parameter
which represents the header field:

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

Since space, double-quote, semicolon, and equals are reserved in URI
header-parameter values, they must all be escaped.  There is also a
consistent problem with excessive indenting (but the RFC Editor will
clean that up).

The first three instances of the 1.1 H-I entry need to have the marked
line amended to match the final instance (which is correct).

   F4 INVITE Example.com -> Carol

   History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302\
+
%3Btext%3D%22Moved%20Temporarily%22>\
                 ;index=1.1;rc=1


   F5 180 Ringing Carol -> Example.com

   History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302\
+
%3Btext%3D%22Moved%20Temporarily%22>\
                 ;index=1.1;rc=1


   F6 INVITE Example.com -> VM

   History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302\
+
%3Btext%3D%22Moved%20Temporarily%22>\
                 ;index=1.1;rc=1


   F7 200 OK VM -> Example.com

   History-Info: <sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302\
                                     %3Btext%3D%22Moved%20Temporarily%22>\
                 ;index=1.1;rc=1

. There are inconsistencies between INVITE F6 and response F7.  Given
the changes from -02 to -03, the I believe the intention is that the
branch 1.2.2 to <sip:vm@example.com;target=sip:carol%40example.com> is
to be a child of the branch 1.2 to <sip:carol@example.com>.  (Which
seems to be the best practice.)  But the change has not been made in
F6.

There is also a ";cause=408" in F6 index 1.3.1 which seems to be
extraneous.  (If it were intended, it would be part of the URI, and
also part of the URI for index 1.3.)

The index value of <sip:vm@192.0.2.5;...> in F7 should be 1.2.2.1.

   F6 INVITE Example.com -> VM

   History-Info: <sip:vm@example.com;target=sip:carol%40example.com>;\
-                      index=1.3;mp=1.2
+                      index=1.2.2;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
+                      ;index=1.2.2.1;rc=1.2.2

   F7 200 OK VM -> Example.com

   History-Info: <sip:vm@example.com;target=sip:carol%40example.com>;\
                       index=1.2.2;mp=1.2
   History-Info: <sip:vm@192.0.2.5;target=sip:carol%40example.com>;\
-                      index=1.2.2;rc=1.2.1
+                      index=1.2.2.1;rc=1.2.2

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

#! /bin/perl

# Check consistency of History-Info headers.

use strict;

my($section, $section_full);
my($message, $message_full);
my($hi);
my(%history);

my(@input) = <>;

while ($_ = shift(@input)) {
    chomp;
    if (($section_full, $section) = /^\s*((\d|\d[.\d]*\d)\. .*?)\s*$/)
    {
        # This picks up the table of contents lines as well as the
        # section headers.
        print "\n";
        print "Section $section_full\n";
        $message = '(none)';
        # Clear the known history.
        %history = ();
    } elsif (($message_full, $message) = /^\s*((F\d+) .*?)\s*$/) {
        # This picks up the headers for individual messages
        print "Message $message_full\n";
    } elsif (($hi) = /^\s*(history-info:.*)$/i) {
        # Pick up a History-Info header.
        # Check for continuation lines.
        while ($hi =~ /\\\s*$/) {
            # Grab the next input line.
            $_ = shift(@input);
            chomp;
            # Check for page break lines and ignore them.
            if (/^\s*$/ ||
                /^Barnes, et al/ ||
                /^\014/ ||
                /^Internet-Draft/) {
                # Ignore all these.
            } else {
                # Append this line to the History-Info header.
                $hi .= "\n" . $_;
            }
        }
        # Now remove the continuations.
        $hi =~ s/\s*\\\s*\n\s*//g;
        # Check to see if the H-I entry has changed.
        my($index) = ($hi =~ /index=([\d.]+)/);
        my($previous) = $history{$index};
        if ($previous && $previous ne $hi) {
            # Add "!" if the change isn't adding a Reason header
    value.
            my($x) = $hi;
            $x =~ s/\?reason=[^>&]*//i;
            # print "A $hi\n$x\n$previous\n";
            print "! " unless $x eq $previous;
            print "Changed $index\nfrom: $previous\nto: $hi\n";
        }
        $history{$index} = $hi;
    }
}

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


Section 1.  Overview
. . . . . . . . . . . . . . . . . . . . . . . . . . .  3

Section 2.  Conventions and Terminology
. . . . . . . . . . . . . . . . .  3

Section 3.  Detailed call flows
. . . . . . . . . . . . . . . . . . . . .  3

Section 3.1.  Sequentially Forking (History-Info in Response)
. . . . .  3

Section 3.2.  History-Info with Privacy Header Field
. . . . . . . . . . 11

Section 3.3.  Privacy for a Specific History-Info Entry
. . . . . . . . 14

Section 3.4.  Automatic Call Distribution
. . . . . . . . . . . . . . . 18

Section 3.5.  Determining the Alias used.
. . . . . . . . . . . . . . . 23

Section 3.6.  PBX Voicemail Example
. . . . . . . . . . . . . . . . . . 26

Section 3.7.  Consumer Voicemail Example
. . . . . . . . . . . . . . . . 31

Section 3.8.  GRUU
. . . . . . . . . . . . . . . . . . . . . . . . . . . 35

Section 3.9.  Limited Use Address
. . . . . . . . . . . . . . . . . . . 38

Section 3.10. Service Invocation
. . . . . . . . . . . . . . . . . . . . 41

Section 3.11. Toll Free Number
. . . . . . . . . . . . . . . . . . . . . 41

Section 4.  Security Considerations
. . . . . . . . . . . . . . . . . . . 44

Section 5.  IANA Considerations
. . . . . . . . . . . . . . . . . . . . . 44

Section 5.1.  Acknowledgements
. . . . . . . . . . . . . . . . . . . . . 44

Section 6.  Informative References
. . . . . . . . . . . . . . . . . . . . 44

Section 1.  Overview

Section 2.  Conventions and Terminology

Section 3.  Detailed call flows

Section 3.1.  Sequentially Forking (History-Info in Response)
Message F1 INVITE alice -> example.com
Message F2 INVITE  example.com -> Bob
Message F3 100 Trying example.com -> alice
Message F4 302 Moved Temporarily Bob -> example.com
Message F5 ACK example.com -> Bob
Message F6 INVITE example.com -> office
Changed 1.1
from: History-Info: <sip:bob@192.0.2.4>;index=1.1;rc=1
to: History-Info:
<sip:bob@192.0.2.4?Reason=SIP%3Bcause%3D302>;index=1.1;rc=1
Message F7 180 Ringing office -> example.com
Message F8 180 Ringing example.com -> alice
! Changed 1.2.1
from: History-Info: <sip:office@192.0.2.5>;index=1.2.1;rc=1.2
to: History-Info:
<sip:office@192.0.2.5;Reason=SIP%3Bcause%3D180>;index=1.2.1;rc=1.2
Message F9 INVITE example.com -> home
! Changed 1.2.1
from: History-Info:
<sip:office@192.0.2.5;Reason=SIP%3Bcause%3D180>;index=1.2.1;rc=1.2
to: History-Info:
<sip:office@192.0.2.5?Reason=SIP%3Bcause%3D408>;index=1.2.1;index=1.2.1;rc=1.2
Message F10 100 Trying home -> example.com
Message F11 486 Busy Here home -> example.com
Message F12 486 Busy Here example.com -> alice
Changed 1.2
from: History-Info: <sip:office@example.com>;index=1.2;mp=1
to: History-Info:
<sip:office@example.com?Reason=SIP%3Bcause%3D408>;index=1.2;mp=1
! Changed 1.2.1
from: History-Info:
<sip:office@192.0.2.5?Reason=SIP%3Bcause%3D408>;index=1.2.1;index=1.2.1;rc=1.2
to: History-Info:
<sip:office@192.0.2.5?Reason=SIP%3Bcause%3D408>;index=1.2.1;rc=1.2
Message F13 ACK example.com -> home
Message F14 ACK alice -> example.com

Section 3.2.  History-Info with Privacy Header Field
Message F1 INVITE alice -> atlanta.example.com
Message F2 INVITE  atlanta.example.com -> biloxi.example.com
! Changed 1
from: History-Info: <sip:bob@biloxi.example.com;p=x>;index=1
to: History-Info: <sip:anonymous@anonymous.invalid>;index=1
Message F3 INVITE  biloxi.example.com -> Bob
Message F4 200 OK  Bob -> biloxi.example.com
Message F5 200 OK  biloxi.example.com -> atlanta.example.com
! Changed 1.1.1
from: History-Info:
<sip:bob@192.0.1.11?Privacy=history>;index=1.1.1;rc=1.1
to: History-Info: <sip:anonymous@anonymous.invalid>;index=1.1.1;rc=1.1
Message F6 200 OK  atlanta.example.com -> Alice

Section 3.3.  Privacy for a Specific History-Info Entry
Message F1 INVITE alice -> atlanta.example.com
Message F2 INVITE  atlanta.example.com -> biloxi.example.com
Message F3 INVITE  biloxi.example.com -> Bob
Message F4 200 OK  Bob -> biloxi.example.com
Message F5 200 OK  biloxi.example.com -> atlanta.example.com
! Changed 1.1.1
from: History-Info:
<sip:bob@192.0.1.11?Privacy=history>;index=1.1.1;rc=1.1
to: History-Info: <sip:anonymous@anonymous.invalid>;index=1.1.1;rc=1.1
Message F6 200 OK  atlanta.example.com -> Alice

Section 3.4.  Automatic Call Distribution
Message F1 INVITE Alice -> Example.com
Message F2 INVITE Example.com -> Gold.Example.com
Message F3 302 Moved Temporarily Gold.Example.com -> Example.com
Message F4 INVITE Example.com -> Silver.Example.com
Changed 1.1
from: History-Info: <sip:Gold@gold.example.com>;rc=1;index=1.1
to: History-Info:
<sip:Gold@gold.example.com?Reason=SIP%3Bcause%3D302>;rc=1;index=1.1
Message F5 INVITE Silver.Example.com -> Agent
! Changed 1.1
from: History-Info:
<sip:Gold@gold.example.com?Reason=SIP%3Bcause%3D302>;rc=1;index=1.1
to: History-Info:
<sip:Gold@gold.example.com?Reason=SIP%3Bcause%3D302>;index=1.1
Message F6 200 OK Agent -> Silver.Example.com
Message F7 200 OK Silver.Example.com -> Example.com
Message F8 200 OK Example.com -> Alice
Message F9 ACK Alice -> Agent

Section 3.5.  Determining the Alias used.
Message F1 REGISTER John -> Example.com
Message F2 200 OK Example.com -> John
Message F3 INVITE Alice -> Example.com
Message F4 INVITE Example.com -> John

Section 3.6.  PBX Voicemail Example
Message F1 INVITE Alice -> Example.com
Message F2 INVITE Example.com -> Bob
Message F3 302 Moved Temporarily Bob -> Example.com
Message F4 INVITE Example.com -> Carol
Changed 1.1
from: History-Info: <sip:bob@192.0.2.5>;index=1.1;rc=1
to: History-Info:
<sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302>;index=1.1;rc=1
Message F5 180 Ringing Carol -> Example.com
Message F6 INVITE Example.com -> VM
! Changed 1.2.1
from: History-Info: <sip:carol@192.0.2.4>;index=1.2.1;rc=1.2
to: History-Info: <sip:carol@192.0.2.4;cause=480>;index=1.2.1;rc=1.2
Message F7 200 OK VM -> Example.com
! Changed 1.2
from: History-Info: <sip:carol@example.com>;index=1.2;mp=1
to: History-Info:
<sip:carol@example.com;cause=480?Reason=SIP%3Bcause%3D408>;index=1.2;mp=1
Changed 1.2.1
from: History-Info: <sip:carol@192.0.2.4;cause=480>;index=1.2.1;rc=1.2
to: History-Info:
<sip:carol@192.0.2.4;cause=480?Reason=SIP%3Bcause%3D408>;index=1.2.1;rc=1.2
! Changed 1.3
from: History-Info:
<sip:vm@example.com;target=sip:bob%40example.com;cause=480>;index=1.3;mp=1
to: History-Info:
<sip:vm@example.com;target=sip:bob%40example.com;cause=408>;index=1.3;mp=1
! Changed 1.3.1
from: History-Info:
<sip:vm@192.0.2.6;target=sip:bob%40example.com;cause=480>;index=1.3.1;rc=1.3
to: History-Info:
<sip:vm@192.0.2.6;target=sip:bob%40example.com;cause=408>;index=1.3.1;rc=1.3

Section 3.7.  Consumer Voicemail Example
Message F1 INVITE Alice -> Example.com
Message F2 INVITE Example.com -> Bob
Message F3 302 Moved Temporarily Bob -> Example.com
Message F4 INVITE Example.com -> Carol
Changed 1.1
from: History-Info: <sip:bob@192.0.2.5>;index=1.1;rc=1
to: History-Info:
<sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302%3Btext%3D"Moved
Temporarily">;index=1.1;rc=1
Message F5 180 Ringing Carol -> Example.com
! Changed 1.1
from: History-Info:
<sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302%3Btext%3D"Moved
Temporarily">;index=1.1;rc=1
to: History-Info:
<sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302;text="Moved
Temporarily">;index=1.1;rc=1
Message F6 INVITE Example.com -> VM
Changed 1.2
from: History-Info: <sip:carol@example.com>;index=1.2;mp=1
to: History-Info:
<sip:carol@example.com?Reason=SIP%3Bcause%3D408>;index=1.2;mp=1
Message F7 200 OK VM -> Example.com
! Changed 1.1
from: History-Info:
<sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302;text="Moved
Temporarily">;index=1.1;rc=1
to: History-Info:
<sip:bob@192.0.2.5?Reason=SIP%3Bcause%3D302%3Btext%3D%22Moved%20Temporarily%22>;index=1.1;rc=1
Changed 1.2.1
from: History-Info: <sip:carol@192.0.2.4>;index=1.2.1;rc=1.2
to: History-Info:
<sip:carol@192.0.2.4?Reason=SIP%3Bcause%3D408>;index=1.2.1;rc=1.2
! Changed 1.2.2
from: History-Info:
<sip:vm@example.com;target=sip:carol%40example.com>;index=1.2.2;mp=1.2
to: History-Info:
<sip:vm@192.0.2.5;target=sip:carol%40example.com>;index=1.2.2;rc=1.2.1

Section 3.8.  GRUU
Message F1 REGISTER John -> Example.com
Message F2 200 OK Example.com -> John
Message F3 INVITE Alice -> Example.com
Message F4 INVITE Example.com -> John

Section 3.9.  Limited Use Address
Message F1 REGISTER John -> Example.com
Message F2 200 OK Example.com -> John
Message F3 INVITE Alice -> Example.com
Message F4 INVITE Example.com -> John

Section 3.10.  Service Invocation

Section 3.11.  Toll Free Number

Section 4.  Security Considerations

Section 5.  IANA Considerations

Section 5.1.  Acknowledgements

Section 6.  Informative References

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

Dale

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To: Dale R. Worley <worley@ariadne.com>
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Cc: sipcore@ietf.org
Subject: Re: [sipcore] Copy edits for draft-ietf-sipcore-rfc4244bis-callflows-03
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Hi Dale;

 Thanks for going through the draft once again.=20

 I will reflect these and update the draft shortly.=20

 Many Thanks
  Shida

On Mar 22, 2013, at 8:57 AM, Dale R. Worley wrote:

> This is a bunch of copy-edits to
> draft-ietf-sipcore-rfc4244bis-callflows-03.  Section pointers are
> marked with "*" at the left margin.  Individual items are marked with
> "." at the left margin.
>=20
> (Finding the inconsistencies between the various History-Info values
> was made infinitely easier by a Perl script, "hi-consistency", which
> is appended.  If you give that script the -03 draft, the output
> flags inconsistent values of the H-I headers within each section of
> the draft.  The output it generates from -03 is also appended.)
>=20
> * All of the figure headings have the "figure N" part doubled:
>=20
>            Figure 1: Figure 1: Example with Sequential Forking
>          Figure 2: Figure 2: Example with Privacy Header Fields
>    Figure 3: Figure 3: Example with Privacy Header Field for Specific
>        Figure 4: Figure 4: Example for Automatic Call Distribution
>                     Figure 5: Figure 5: Alias Example
>             Figure 6: Figure 6: Enterprise Voivemail Example
>              Figure 7: Figure 7: Consumer Voivemail Example
>                     Figure 8: Figure 8: GRUU Example
>              Figure 9: Figure 9: Limited Use Address Example
>               Figure 10: Figure 10: Service Number Example
>=20
> (I expect the RFC Editor can correct this.)
>=20
> * Section 3.1
>=20
> . =46rom F8, I see that the rule is to record 1xx responses as well as
> final failure responses:
>=20
>       History-Info: <sip:office@192.0.2.5;Reason=3DSIP%3Bcause%3D180>;\
>                                                         =
index=3D1.2.1;rc=3D1.2
>=20
> (If recording 1xx responses with Reason is not intended, please modify
> the following items by removing any Reason header-value that specifies
> a 1xx cause value.)
>=20
> . Correct the punctuation:
>=20
>       F8 180 Ringing example.com -> alice
>=20
>    -  History-Info: <sip:office@192.0.2.5;Reason=3DSIP%3Bcause%3D180>;\
>    +  History-Info: <sip:office@192.0.2.5?Reason=3DSIP%3Bcause%3D180>;\
>                                                         =
index=3D1.2.1;rc=3D1.2
>=20
> . Remove duplicate index parameter:
>=20
>       F9 INVITE example.com -> home
>=20
>       History-Info: <sip:office@192.0.2.5?Reason=3DSIP%3Bcause%3D408>;\
>    -                index=3D1.2.1;index=3D1.2.1;rc=3D1.2
>    +                index=3D1.2.1;rc=3D1.2
>=20
> . The handling of the Reason on index=3D1.2 seems to be inconsistent:
>=20
> Branch 1.2 is first seen in INVITE F6.  (INVITE F6 itself is
> index=3D1.2.1, generated by internal forking from branch 1.2.)
>=20
>   F6 INVITE example.com -> office
>=20
>   History-Info: <sip:office@example.com>;index=3D1.2;mp=3D1
>=20
> INVITE F9 is branch 1.3.1, generated by internal forking from branch
> 1.3, which was generated due to the failure of INVITE F6 (by the
> timeout between F8 and F9).  In F9, index=3D1.2.1 is shown with a =
Reason
> value (which is mandatory), but index=3D1.2 is not (which is allowed,
> but not best practice).
>=20
>   F9 INVITE example.com -> home
>=20
>   History-Info: <sip:office@example.com>;index=3D1.2;mp=3D1
>   History-Info: <sip:office@192.0.2.5?Reason=3DSIP%3Bcause%3D408>;\
>                 index=3D1.2.1;index=3D1.2.1;rc=3D1.2
>=20
> INVITE F11 follows the pattern of F9.
>=20
>   F11 486 Busy Here home -> example.com
>=20
>   History-Info: <sip:office@example.com>;index=3D1.2;mp=3D1
>=20
> But response F12 shows the Reason value on index=3D1.2 that is best
> practice.
>=20
>   F12 486 Busy Here example.com -> alice
>=20
>   History-Info: <sip:office@example.com?Reason=3DSIP%3Bcause%3D408>;\
>                                                           =
index=3D1.2;mp=3D1
>=20
> I think what is intended is for F8, F9 and F11 to have the Reason
> value on
> index=3D1.2 as well.
>=20
>       F8 180 Ringing example.com -> alice
>=20
>    -  History-Info: <sip:office@example.com>;index=3D1.2;mp=3D1
>    +  History-Info:
>    <sip:office@example.com?Reason=3DSIP%3Bcause%3D180>;\
>    +                                                  index=3D1.2;mp=3D1=

>=20
>       F9 INVITE example.com -> home
>=20
>    -  History-Info: <sip:office@example.com>;index=3D1.2;mp=3D1
>    +  History-Info:
>    <sip:office@example.com?Reason=3DSIP%3Bcause%3D408>;\
>    +                                                  index=3D1.2;mp=3D1=

>=20
>       F11 486 Busy Here home -> example.com
>=20
>    -  History-Info: <sip:office@example.com>;index=3D1.2;mp=3D1
>    +  History-Info:
>    <sip:office@example.com?Reason=3DSIP%3Bcause%3D408>;\
>    +                                                  index=3D1.2;mp=3D1=

>=20
> * Section 3.4
>=20
> . Make F5 consistent with preceding messages:
>=20
>       F5 INVITE Silver.Example.com -> Agent
>=20
>       History-Info:
>       <sip:Gold@gold.example.com?Reason=3DSIP%3Bcause%3D302>;\
>    -                  index=3D1.1
>    +                  rc=3D1;index=3D1.1
>=20
> Same for F6, F7, and F8.
>=20
> * Section 3.6
>=20
> . URIs for 1.2 and 1.2.1 are not consistent:
>=20
>   F4 INVITE Example.com -> Carol
>=20
>   History-Info: <sip:carol@example.com>;index=3D1.2;mp=3D1
>   History-Info: <sip:carol@192.0.2.4>;index=3D1.2.1;rc=3D1.2
>=20
>   F5 180 Ringing Carol -> Example.com
>=20
>   History-Info: <sip:carol@example.com>;index=3D1.2;mp=3D1
>   History-Info: <sip:carol@192.0.2.4>;index=3D1.2.1;rc=3D1.2
>=20
>   F6 INVITE Example.com -> VM
>=20
>   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
>=20
>   F7 200 OK VM -> Example.com
>=20
>   History-Info:
>   <sip:carol@example.com;cause=3D480?Reason=3DSIP%3Bcause%3D\
>                      408>;index=3D1.2;mp=3D1
>   History-Info: <sip:carol@192.0.2.4;cause=3D480?Reason=3DSIP%3Bcause%3D=
\
>                      408>;index=3D1.2.1;rc=3D1.2
>=20
> My understanding is that the intention is to show use of the cause URI
> parameter (not the cause value in Reason!) per RFC 4458.  (See
> Marianne Mohali's explanation of the situation at
> http://www.ietf.org/mail-archive/web/sipcore/current/msg05514.html.)
> If that is so, the URI for branch 1.2 would be generated as shown in
> F7, <sip:carol@example.com;cause=3D480>, and presumably the parameter
> would be carried into the URI for branch 1.2.1.  That requires the
> changes shown below.
>=20
> . Also, in INVITE F6, the timeout (408) response to branches 1.2
> and 1.2.1 would already be recorded, as they are in response F7 (which
> is generated from INVITE F6!).  Those changes are also included below.
>=20
> . Also, in response F7, the cause URI parameter recorded in branches
> 1.3 and 1.3.1 is not the same as in INVITE F6:  F6 has "cause=3D480" =
and
> F7 has "cause=3D408".  Which code to choose depends on the application
> logic (see the discussion around Marianne Mohali's message referenced
> above), and I have no particular opinion one way or the other, but the
> two messages must be consistent.  No change for that is included
> below.
>=20
>       F4 INVITE Example.com -> Carol
>=20
>    -  History-Info: <sip:carol@example.com>;index=3D1.2;mp=3D1
>    -  History-Info: <sip:carol@192.0.2.4>;index=3D1.2.1;rc=3D1.2
>    +  History-Info: <sip:carol@example.com;cause=3D480>;index=3D1.2;mp=3D=
1
>    +  History-Info:
>    -  <sip:carol@192.0.2.4;cause=3D480>;index=3D1.2.1;rc=3D1.2
>=20
>       F5 180 Ringing Carol -> Example.com
>=20
>    -  History-Info: <sip:carol@example.com>;index=3D1.2;mp=3D1
>    -  History-Info: <sip:carol@192.0.2.4>;index=3D1.2.1;rc=3D1.2
>    +  History-Info: <sip:carol@example.com;cause=3D480>;index=3D1.2;mp=3D=
1
>    +  History-Info:
>    -  <sip:carol@192.0.2.4;cause=3D480>;index=3D1.2.1;rc=3D1.2
>=20
>       F6 INVITE Example.com -> VM
>=20
>    -  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:carol@example.com;cause=3D480?Reason=3DSIP%3Bcause%3D\
>    +                     408>;index=3D1.2;mp=3D1
>    +  History-Info:
>    -  <sip:carol@192.0.2.4;cause=3D480?Reason=3DSIP%3Bcause%3D\
>    +                     408>;index=3D1.2.1;rc=3D1.2
>=20
>       F7 200 OK VM -> Example.com
>=20
>       History-Info:
>       <sip:carol@example.com;cause=3D480?Reason=3DSIP%3Bcause%3D\
>                          408>;index=3D1.2;mp=3D1
>       History-Info:
>       <sip:carol@192.0.2.4;cause=3D480?Reason=3DSIP%3Bcause%3D\
>                          408>;index=3D1.2.1;rc=3D1.2
>=20
> * Section 3.7
>=20
> . There are a number of H-I entries that want to encode a cause text
> "Moved Temporarily".  More exactly, they encode a URI header parameter
> which represents the header field:
>=20
>    Reason: SIP;cause=3D302;text=3D"Moved Temporarily"
>=20
> Since space, double-quote, semicolon, and equals are reserved in URI
> header-parameter values, they must all be escaped.  There is also a
> consistent problem with excessive indenting (but the RFC Editor will
> clean that up).
>=20
> The first three instances of the 1.1 H-I entry need to have the marked
> line amended to match the final instance (which is correct).
>=20
>   F4 INVITE Example.com -> Carol
>=20
>   History-Info: <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302\
> +
> %3Btext%3D%22Moved%20Temporarily%22>\
>                 ;index=3D1.1;rc=3D1
>=20
>=20
>   F5 180 Ringing Carol -> Example.com
>=20
>   History-Info: <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302\
> +
> %3Btext%3D%22Moved%20Temporarily%22>\
>                 ;index=3D1.1;rc=3D1
>=20
>=20
>   F6 INVITE Example.com -> VM
>=20
>   History-Info: <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302\
> +
> %3Btext%3D%22Moved%20Temporarily%22>\
>                 ;index=3D1.1;rc=3D1
>=20
>=20
>   F7 200 OK VM -> Example.com
>=20
>   History-Info: <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302\
>                                     =
%3Btext%3D%22Moved%20Temporarily%22>\
>                 ;index=3D1.1;rc=3D1
>=20
> . There are inconsistencies between INVITE F6 and response F7.  Given
> the changes from -02 to -03, the I believe the intention is that the
> branch 1.2.2 to <sip:vm@example.com;target=3Dsip:carol%40example.com> =
is
> to be a child of the branch 1.2 to <sip:carol@example.com>.  (Which
> seems to be the best practice.)  But the change has not been made in
> F6.
>=20
> There is also a ";cause=3D408" in F6 index 1.3.1 which seems to be
> extraneous.  (If it were intended, it would be part of the URI, and
> also part of the URI for index 1.3.)
>=20
> The index value of <sip:vm@192.0.2.5;...> in F7 should be 1.2.2.1.
>=20
>   F6 INVITE Example.com -> VM
>=20
>   History-Info: <sip:vm@example.com;target=3Dsip:carol%40example.com>;\
> -                      index=3D1.3;mp=3D1.2
> +                      index=3D1.2.2;mp=3D1.2
>   History-Info: <sip:vm@192.0.2.5;target=3Dsip:carol%40example.com>\
> -                      ;cause=3D408;index=3D1.3.1;rc=3D1.3
> +                      ;index=3D1.2.2.1;rc=3D1.2.2
>=20
>   F7 200 OK VM -> Example.com
>=20
>   History-Info: <sip:vm@example.com;target=3Dsip:carol%40example.com>;\
>                       index=3D1.2.2;mp=3D1.2
>   History-Info: <sip:vm@192.0.2.5;target=3Dsip:carol%40example.com>;\
> -                      index=3D1.2.2;rc=3D1.2.1
> +                      index=3D1.2.2.1;rc=3D1.2.2
>=20
> ----------------------------------------------------------------------
>=20
> #! /bin/perl
>=20
> # Check consistency of History-Info headers.
>=20
> use strict;
>=20
> my($section, $section_full);
> my($message, $message_full);
> my($hi);
> my(%history);
>=20
> my(@input) =3D <>;
>=20
> while ($_ =3D shift(@input)) {
>    chomp;
>    if (($section_full, $section) =3D /^\s*((\d|\d[.\d]*\d)\. =
.*?)\s*$/)
>    {
>        # This picks up the table of contents lines as well as the
>        # section headers.
>        print "\n";
>        print "Section $section_full\n";
>        $message =3D '(none)';
>        # Clear the known history.
>        %history =3D ();
>    } elsif (($message_full, $message) =3D /^\s*((F\d+) .*?)\s*$/) {
>        # This picks up the headers for individual messages
>        print "Message $message_full\n";
>    } elsif (($hi) =3D /^\s*(history-info:.*)$/i) {
>        # Pick up a History-Info header.
>        # Check for continuation lines.
>        while ($hi =3D~ /\\\s*$/) {
>            # Grab the next input line.
>            $_ =3D shift(@input);
>            chomp;
>            # Check for page break lines and ignore them.
>            if (/^\s*$/ ||
>                /^Barnes, et al/ ||
>                /^\014/ ||
>                /^Internet-Draft/) {
>                # Ignore all these.
>            } else {
>                # Append this line to the History-Info header.
>                $hi .=3D "\n" . $_;
>            }
>        }
>        # Now remove the continuations.
>        $hi =3D~ s/\s*\\\s*\n\s*//g;
>        # Check to see if the H-I entry has changed.
>        my($index) =3D ($hi =3D~ /index=3D([\d.]+)/);
>        my($previous) =3D $history{$index};
>        if ($previous && $previous ne $hi) {
>            # Add "!" if the change isn't adding a Reason header
>    value.
>            my($x) =3D $hi;
>            $x =3D~ s/\?reason=3D[^>&]*//i;
>            # print "A $hi\n$x\n$previous\n";
>            print "! " unless $x eq $previous;
>            print "Changed $index\nfrom: $previous\nto: $hi\n";
>        }
>        $history{$index} =3D $hi;
>    }
> }
>=20
> ----------------------------------------------------------------------
>=20
>=20
> Section 1.  Overview
> . . . . . . . . . . . . . . . . . . . . . . . . . . .  3
>=20
> Section 2.  Conventions and Terminology
> . . . . . . . . . . . . . . . . .  3
>=20
> Section 3.  Detailed call flows
> . . . . . . . . . . . . . . . . . . . . .  3
>=20
> Section 3.1.  Sequentially Forking (History-Info in Response)
> . . . . .  3
>=20
> Section 3.2.  History-Info with Privacy Header Field
> . . . . . . . . . . 11
>=20
> Section 3.3.  Privacy for a Specific History-Info Entry
> . . . . . . . . 14
>=20
> Section 3.4.  Automatic Call Distribution
> . . . . . . . . . . . . . . . 18
>=20
> Section 3.5.  Determining the Alias used.
> . . . . . . . . . . . . . . . 23
>=20
> Section 3.6.  PBX Voicemail Example
> . . . . . . . . . . . . . . . . . . 26
>=20
> Section 3.7.  Consumer Voicemail Example
> . . . . . . . . . . . . . . . . 31
>=20
> Section 3.8.  GRUU
> . . . . . . . . . . . . . . . . . . . . . . . . . . . 35
>=20
> Section 3.9.  Limited Use Address
> . . . . . . . . . . . . . . . . . . . 38
>=20
> Section 3.10. Service Invocation
> . . . . . . . . . . . . . . . . . . . . 41
>=20
> Section 3.11. Toll Free Number
> . . . . . . . . . . . . . . . . . . . . . 41
>=20
> Section 4.  Security Considerations
> . . . . . . . . . . . . . . . . . . . 44
>=20
> Section 5.  IANA Considerations
> . . . . . . . . . . . . . . . . . . . . . 44
>=20
> Section 5.1.  Acknowledgements
> . . . . . . . . . . . . . . . . . . . . . 44
>=20
> Section 6.  Informative References
> . . . . . . . . . . . . . . . . . . . . 44
>=20
> Section 1.  Overview
>=20
> Section 2.  Conventions and Terminology
>=20
> Section 3.  Detailed call flows
>=20
> Section 3.1.  Sequentially Forking (History-Info in Response)
> Message F1 INVITE alice -> example.com
> Message F2 INVITE  example.com -> Bob
> Message F3 100 Trying example.com -> alice
> Message F4 302 Moved Temporarily Bob -> example.com
> Message F5 ACK example.com -> Bob
> Message F6 INVITE example.com -> office
> Changed 1.1
> from: History-Info: <sip:bob@192.0.2.4>;index=3D1.1;rc=3D1
> to: History-Info:
> <sip:bob@192.0.2.4?Reason=3DSIP%3Bcause%3D302>;index=3D1.1;rc=3D1
> Message F7 180 Ringing office -> example.com
> Message F8 180 Ringing example.com -> alice
> ! Changed 1.2.1
> from: History-Info: <sip:office@192.0.2.5>;index=3D1.2.1;rc=3D1.2
> to: History-Info:
> <sip:office@192.0.2.5;Reason=3DSIP%3Bcause%3D180>;index=3D1.2.1;rc=3D1.2=

> Message F9 INVITE example.com -> home
> ! Changed 1.2.1
> from: History-Info:
> <sip:office@192.0.2.5;Reason=3DSIP%3Bcause%3D180>;index=3D1.2.1;rc=3D1.2=

> to: History-Info:
> =
<sip:office@192.0.2.5?Reason=3DSIP%3Bcause%3D408>;index=3D1.2.1;index=3D1.=
2.1;rc=3D1.2
> Message F10 100 Trying home -> example.com
> Message F11 486 Busy Here home -> example.com
> Message F12 486 Busy Here example.com -> alice
> Changed 1.2
> from: History-Info: <sip:office@example.com>;index=3D1.2;mp=3D1
> to: History-Info:
> <sip:office@example.com?Reason=3DSIP%3Bcause%3D408>;index=3D1.2;mp=3D1
> ! Changed 1.2.1
> from: History-Info:
> =
<sip:office@192.0.2.5?Reason=3DSIP%3Bcause%3D408>;index=3D1.2.1;index=3D1.=
2.1;rc=3D1.2
> to: History-Info:
> <sip:office@192.0.2.5?Reason=3DSIP%3Bcause%3D408>;index=3D1.2.1;rc=3D1.2=

> Message F13 ACK example.com -> home
> Message F14 ACK alice -> example.com
>=20
> Section 3.2.  History-Info with Privacy Header Field
> Message F1 INVITE alice -> atlanta.example.com
> Message F2 INVITE  atlanta.example.com -> biloxi.example.com
> ! Changed 1
> from: History-Info: <sip:bob@biloxi.example.com;p=3Dx>;index=3D1
> to: History-Info: <sip:anonymous@anonymous.invalid>;index=3D1
> Message F3 INVITE  biloxi.example.com -> Bob
> Message F4 200 OK  Bob -> biloxi.example.com
> Message F5 200 OK  biloxi.example.com -> atlanta.example.com
> ! Changed 1.1.1
> from: History-Info:
> <sip:bob@192.0.1.11?Privacy=3Dhistory>;index=3D1.1.1;rc=3D1.1
> to: History-Info: <sip:anonymous@anonymous.invalid>;index=3D1.1.1;rc=3D1=
.1
> Message F6 200 OK  atlanta.example.com -> Alice
>=20
> Section 3.3.  Privacy for a Specific History-Info Entry
> Message F1 INVITE alice -> atlanta.example.com
> Message F2 INVITE  atlanta.example.com -> biloxi.example.com
> Message F3 INVITE  biloxi.example.com -> Bob
> Message F4 200 OK  Bob -> biloxi.example.com
> Message F5 200 OK  biloxi.example.com -> atlanta.example.com
> ! Changed 1.1.1
> from: History-Info:
> <sip:bob@192.0.1.11?Privacy=3Dhistory>;index=3D1.1.1;rc=3D1.1
> to: History-Info: <sip:anonymous@anonymous.invalid>;index=3D1.1.1;rc=3D1=
.1
> Message F6 200 OK  atlanta.example.com -> Alice
>=20
> Section 3.4.  Automatic Call Distribution
> Message F1 INVITE Alice -> Example.com
> Message F2 INVITE Example.com -> Gold.Example.com
> Message F3 302 Moved Temporarily Gold.Example.com -> Example.com
> Message F4 INVITE Example.com -> Silver.Example.com
> Changed 1.1
> from: History-Info: <sip:Gold@gold.example.com>;rc=3D1;index=3D1.1
> to: History-Info:
> <sip:Gold@gold.example.com?Reason=3DSIP%3Bcause%3D302>;rc=3D1;index=3D1.=
1
> Message F5 INVITE Silver.Example.com -> Agent
> ! Changed 1.1
> from: History-Info:
> <sip:Gold@gold.example.com?Reason=3DSIP%3Bcause%3D302>;rc=3D1;index=3D1.=
1
> to: History-Info:
> <sip:Gold@gold.example.com?Reason=3DSIP%3Bcause%3D302>;index=3D1.1
> Message F6 200 OK Agent -> Silver.Example.com
> Message F7 200 OK Silver.Example.com -> Example.com
> Message F8 200 OK Example.com -> Alice
> Message F9 ACK Alice -> Agent
>=20
> Section 3.5.  Determining the Alias used.
> Message F1 REGISTER John -> Example.com
> Message F2 200 OK Example.com -> John
> Message F3 INVITE Alice -> Example.com
> Message F4 INVITE Example.com -> John
>=20
> Section 3.6.  PBX Voicemail Example
> Message F1 INVITE Alice -> Example.com
> Message F2 INVITE Example.com -> Bob
> Message F3 302 Moved Temporarily Bob -> Example.com
> Message F4 INVITE Example.com -> Carol
> Changed 1.1
> from: History-Info: <sip:bob@192.0.2.5>;index=3D1.1;rc=3D1
> to: History-Info:
> <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302>;index=3D1.1;rc=3D1
> Message F5 180 Ringing Carol -> Example.com
> Message F6 INVITE Example.com -> VM
> ! Changed 1.2.1
> from: History-Info: <sip:carol@192.0.2.4>;index=3D1.2.1;rc=3D1.2
> to: History-Info: <sip:carol@192.0.2.4;cause=3D480>;index=3D1.2.1;rc=3D1=
.2
> Message F7 200 OK VM -> Example.com
> ! Changed 1.2
> from: History-Info: <sip:carol@example.com>;index=3D1.2;mp=3D1
> to: History-Info:
> =
<sip:carol@example.com;cause=3D480?Reason=3DSIP%3Bcause%3D408>;index=3D1.2=
;mp=3D1
> Changed 1.2.1
> from: History-Info: <sip:carol@192.0.2.4;cause=3D480>;index=3D1.2.1;rc=3D=
1.2
> to: History-Info:
> =
<sip:carol@192.0.2.4;cause=3D480?Reason=3DSIP%3Bcause%3D408>;index=3D1.2.1=
;rc=3D1.2
> ! Changed 1.3
> from: History-Info:
> =
<sip:vm@example.com;target=3Dsip:bob%40example.com;cause=3D480>;index=3D1.=
3;mp=3D1
> to: History-Info:
> =
<sip:vm@example.com;target=3Dsip:bob%40example.com;cause=3D408>;index=3D1.=
3;mp=3D1
> ! Changed 1.3.1
> from: History-Info:
> =
<sip:vm@192.0.2.6;target=3Dsip:bob%40example.com;cause=3D480>;index=3D1.3.=
1;rc=3D1.3
> to: History-Info:
> =
<sip:vm@192.0.2.6;target=3Dsip:bob%40example.com;cause=3D408>;index=3D1.3.=
1;rc=3D1.3
>=20
> Section 3.7.  Consumer Voicemail Example
> Message F1 INVITE Alice -> Example.com
> Message F2 INVITE Example.com -> Bob
> Message F3 302 Moved Temporarily Bob -> Example.com
> Message F4 INVITE Example.com -> Carol
> Changed 1.1
> from: History-Info: <sip:bob@192.0.2.5>;index=3D1.1;rc=3D1
> to: History-Info:
> <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302%3Btext%3D"Moved
> Temporarily">;index=3D1.1;rc=3D1
> Message F5 180 Ringing Carol -> Example.com
> ! Changed 1.1
> from: History-Info:
> <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302%3Btext%3D"Moved
> Temporarily">;index=3D1.1;rc=3D1
> to: History-Info:
> <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302;text=3D"Moved
> Temporarily">;index=3D1.1;rc=3D1
> Message F6 INVITE Example.com -> VM
> Changed 1.2
> from: History-Info: <sip:carol@example.com>;index=3D1.2;mp=3D1
> to: History-Info:
> <sip:carol@example.com?Reason=3DSIP%3Bcause%3D408>;index=3D1.2;mp=3D1
> Message F7 200 OK VM -> Example.com
> ! Changed 1.1
> from: History-Info:
> <sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302;text=3D"Moved
> Temporarily">;index=3D1.1;rc=3D1
> to: History-Info:
> =
<sip:bob@192.0.2.5?Reason=3DSIP%3Bcause%3D302%3Btext%3D%22Moved%20Temporar=
ily%22>;index=3D1.1;rc=3D1
> Changed 1.2.1
> from: History-Info: <sip:carol@192.0.2.4>;index=3D1.2.1;rc=3D1.2
> to: History-Info:
> <sip:carol@192.0.2.4?Reason=3DSIP%3Bcause%3D408>;index=3D1.2.1;rc=3D1.2
> ! Changed 1.2.2
> from: History-Info:
> <sip:vm@example.com;target=3Dsip:carol%40example.com>;index=3D1.2.2;mp=3D=
1.2
> to: History-Info:
> <sip:vm@192.0.2.5;target=3Dsip:carol%40example.com>;index=3D1.2.2;rc=3D1=
.2.1
>=20
> Section 3.8.  GRUU
> Message F1 REGISTER John -> Example.com
> Message F2 200 OK Example.com -> John
> Message F3 INVITE Alice -> Example.com
> Message F4 INVITE Example.com -> John
>=20
> Section 3.9.  Limited Use Address
> Message F1 REGISTER John -> Example.com
> Message F2 200 OK Example.com -> John
> Message F3 INVITE Alice -> Example.com
> Message F4 INVITE Example.com -> John
>=20
> Section 3.10.  Service Invocation
>=20
> Section 3.11.  Toll Free Number
>=20
> Section 4.  Security Considerations
>=20
> Section 5.  IANA Considerations
>=20
> Section 5.1.  Acknowledgements
>=20
> Section 6.  Informative References
>=20
> ----------------------------------------------------------------------
>=20
> Dale
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore

