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


	Title           : Indication of support for keep-alive
	Author(s)       : C. Holmberg
	Filename        : draft-ietf-sipcore-keep-01.txt
	Pages           : 11
	Date            : 2009-12-01

This specification defines a new SIP Via header parameter, "keep"
which SIP entities can use to indicate support of the NAT keep-alive
techniques defined in SIP Outbound, in cases where the Outbound
procedures are not supported or cannot be applied.

Status of this Memo

This Internet-Draft is submitted to IETF in full conformance with the
provisions of BCP 78 and BCP 79.

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From christer.holmberg@ericsson.com  Tue Dec  1 03:18:32 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
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Hi,

I've submitted a new version of the keep draft. There are no changes
from the previous version, but it had expired.

I will produce a new version, with new content, once we get the info
issues resolved - it has really taken more time than expected...=20

A new version of the info draft will be submitted tomorrow.

Regards,

Christer

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

<P><FONT SIZE=3D2 FACE=3D"Arial">I've submitted a new version of the =
keep draft. There are no changes from the previous version, but it had =
expired.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">I will produce a new version, with new =
content, once we get the info issues resolved - it has really taken more =
time than expected... </FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">A new version of the info draft will be =
submitted tomorrow.</FONT>
</P>

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

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

=20

Going over RFC 5626, it looks like a superset of rules defined in
connection-reuse draft. The RFC enhanced the basic feature for UDP, and
added a keep alive mechanism, but all and all It looks like they both
describe 2 different features describing means to use an open connection
for multiple network-transactions from the server side.

=20

What are the future plans for this draft? Will it be canceled due to the
RFC existence? If not, any views on how those 2 features co-exist on a
UE? What if an alias parameter contradicts a flow parameter?

=20

Thanks,

Ofer

=20

_______________________________________________

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

=20

P Before printing this email, kindly think about the environment


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<div class=3DSection1>

<p class=3DMsoNormal>Hi All.<o:p></o:p></p>

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

<p class=3DMsoNormal>Going over RFC 5626, it looks like a superset of =
rules
defined in connection-reuse draft. The RFC enhanced the basic feature =
for UDP,
and added a keep alive mechanism, but all and all It looks like they =
both
describe 2 different features describing means to use an open connection =
for
multiple network-transactions from the server side.<o:p></o:p></p>

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

<p class=3DMsoNormal>What are the future plans for this draft? Will it =
be
canceled due to the RFC existence? If not, any views on how those 2 =
features
co-exist on a UE? What if an alias parameter contradicts a flow =
parameter?<o:p></o:p></p>

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

<p class=3DMsoNormal>Thanks,<o:p></o:p></p>

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

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

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>_____________=
__________________________________</span><span
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'><o:p></o:p></span></p>

<div>

<p class=3DMsoNormal><i><span =
style=3D'font-size:10.0pt;font-family:"Verdana","sans-serif";
color:#707070'>If you lend someone $20, and never see that person again, =
it was
probably worth the investment. </span></i><i><span =
style=3D'font-size:10.0pt;
font-family:"Arial","sans-serif";color:gray'><o:p></o:p></span></i></p>

</div>

<div>

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

</div>

<p class=3DMsoNormal><b><i><span lang=3DEN-GB =
style=3D'font-size:18.0pt;font-family:
Webdings;color:green'>P</span></i></b><b><i><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:"Tahoma","sans-serif";color:green'> =
</span></i></b><b><i><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:green'>=
B</span></i></b><b><i><span
style=3D'font-size:7.5pt;font-family:"Tahoma","sans-serif";color:green'>e=
fore</span></i></b><b><i><span
style=3D'font-size:7.5pt;font-family:"Tahoma","sans-serif";color:green'> =
printing
this email, kindly think&nbsp;about the =
environment</span></i></b><o:p></o:p></p>

</div>

</body>

</html>

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Ofer Goren wrote:
> Going over RFC 5626, it looks like a superset of rules defined in 
> connection-reuse draft. The RFC enhanced the basic feature for UDP, and 
> added a keep alive mechanism, but all and all It looks like they both 
> describe 2 different features describing means to use an open connection 
> for multiple network-transactions from the server side.

Ofer: The intersection of sip-outbound (now rfc5626) and
sip-connect-reuse is detailed in Section 2 of the sip-connect-reuse
draft.

> What are the future plans for this draft?

sip-connect-reuse is in the RFC editor's queue.  It is
currently blocked on sip-domain-certs reaching IESG since
there is a dependency on sip-domain-certs.

Thanks,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
Web:   http://ect.bell-labs.com/who/vkg/

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On Dec 1, 2009, at 6:54 , Vijay K. Gurbani wrote:

> Ofer Goren wrote:
>> Going over RFC 5626, it looks like a superset of rules defined in =
connection-reuse draft. The RFC enhanced the basic feature for UDP, and =
added a keep alive mechanism, but all and all It looks like they both =
describe 2 different features describing means to use an open connection =
for multiple network-transactions from the server side.
>=20
> Ofer: The intersection of sip-outbound (now rfc5626) and
> sip-connect-reuse is detailed in Section 2 of the sip-connect-reuse
> draft.

Another way to view the place under the sun for 5626 vs connect-reuse is =
that 5626 is typically going to be used between a UA and a =
proxy/registrar, while connect-reuse is expected to be used between =
proxies (or with servers such as voicemail servers, conference bridges =
and so forth where.



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Thank you all.

Ofer.

-----Original Message-----
From: Francois Audet [mailto:audet.francois@gmail.com] 
Sent: Tuesday, December 01, 2009 18:50
To: Vijay K. Gurbani
Cc: Ofer Goren; sipcore@ietf.org
Subject: Re: [sipcore] rfc 5626 and connection-reuse-draft


On Dec 1, 2009, at 6:54 , Vijay K. Gurbani wrote:

> Ofer Goren wrote:
>> Going over RFC 5626, it looks like a superset of rules defined in
connection-reuse draft. The RFC enhanced the basic feature for UDP, and
added a keep alive mechanism, but all and all It looks like they both
describe 2 different features describing means to use an open connection
for multiple network-transactions from the server side.
> 
> Ofer: The intersection of sip-outbound (now rfc5626) and
> sip-connect-reuse is detailed in Section 2 of the sip-connect-reuse
> draft.

Another way to view the place under the sun for 5626 vs connect-reuse is
that 5626 is typically going to be used between a UA and a
proxy/registrar, while connect-reuse is expected to be used between
proxies (or with servers such as voicemail servers, conference bridges
and so forth where.




From audet.francois@gmail.com  Tue Dec  1 19:56:09 2009
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That seems pretty reasonable to me. But...

Perhaps we can clarify a bit more in explaining why a voicemail system  
would choose the first versus the last redirecting target.  This is  
typically not a "configuration" as described in your example below  
(although, it is certainly possible).

Typically, a voicemail system will choose the first one IF it is  
responsible for the mailbox of that target. Obviously, it would be  
impossible otherwise. In this latter case, the voicemail system would  
then use the last redirecting number.

This is why I don't think we can put "normative" text, but instead  
"use case examples" as described above. A implementation could decide  
to always use the last redirecting number.

On Nov 30, 2009, at 17:39 , Shida Schubert wrote:

>
> Francois;
>
> This was something Cullen asked we add to the draft too.
>
> I am assuming what we need is to add an explicit text that aids
> voicemail server to identify the target of the voicemail box.
>
> If a call was initially received at Alice then diverted
> to Carol,  then to Bob which then diverts to voicemail,
> the question is for which AoR should the voicemail server
> leave a voicemail?
>
> I think the answer depends on the configuration
> of the voicemail server. But with the "mp" tag and
> "mp" index we can identify when the mapping
> (diversion) took place making it much more
> deterministic to identify the target of the voicemail box.
>
> Looking at the example from the draft below;
>
>  History-Info: <sip:bob@example.com>;index=1
>  History-Info: <sip:bob@192.0.2.3?Reason=SIP;cause=302>;\
>                   index=1.1;rc
>     History-Info: <sip:carol@example.com>;index=1.2;mp=1
>     History-Info: <sip:carol@192.0.2.4>;index=1.2.1;rc
>     History-Info: <sip:vm@example.com>;index=1.3;mp=1.2
>     History-Info: <sip:vm@192.0.2.5>;index=1.3.1
>
> If the voicemail server is configured to use the voicemail box
> of the initial target it can look for the first mp tag and look
> at the hi-entry that mp index references (bob@example.com).
>
> If the voicemail server is configured to use the voicemail box
> of the last target then it can look for the last mp tag and look
> at the hi-entry that mp index references (carol@example.com) etc.
>
> Regards
>  Shida
>
> On Dec 1, 2009, at 7:56 AM, Francois AUDET wrote:
>
>>
>> On Nov 30, 2009, at 5:13 , Alan Johnston wrote:
>>
>>>
>>> Does this document have a good voicemail example?  This is the  
>>> application I am most interested in, so it would be nice to have a  
>>> very clear example of this.
>>>
>>
>> That was the intent of section B.2 (in the appendix).
>>
>> Are you saying you'd want a better voicemail example? Or that we  
>> add more voicemail examples? What do you have in mind?
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>


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From: Francois Audet <audet.francois@gmail.com>
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Subject: Re: [sipcore] I-D Action:draft-ietf-sipcore-keep-01.txt
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Should update reference to outbound to RFC 5626.

On Dec 1, 2009, at 03:15 , Internet-Drafts@ietf.org wrote:

> A New Internet-Draft is available from the on-line Internet-Drafts  
> directories.
> This draft is a work item of the Session Initiation Protocol Core  
> Working Group of the IETF.
>
>
> 	Title           : Indication of support for keep-alive
> 	Author(s)       : C. Holmberg
> 	Filename        : draft-ietf-sipcore-keep-01.txt
> 	Pages           : 11
> 	Date            : 2009-12-01
>
> This specification defines a new SIP Via header parameter, "keep"
> which SIP entities can use to indicate support of the NAT keep-alive
> techniques defined in SIP Outbound, in cases where the Outbound
> procedures are not supported or cannot be applied.
>
> Status of this Memo
>
> This Internet-Draft is submitted to IETF in full conformance with the
> provisions of BCP 78 and BCP 79.
>
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From shida@ntt-at.com  Tue Dec  1 21:28:34 2009
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Cc: sipcore@ietf.org
Subject: Re: [sipcore] Some comments on draft-barnes-sipcore-rfc4244bis-03
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 I don't think we need normative text, the whole examples are=20
in appendix anyways, but I think what people want is to have=20
explicit text on how one can determine the first versus the=20
last redirecting target. I am not the one asking for them so=20
I can't speak for them though..=20

 Regards
  Shida

On Dec 2, 2009, at 12:55 PM, Francois Audet wrote:

> That seems pretty reasonable to me. But...
>=20
> Perhaps we can clarify a bit more in explaining why a voicemail system =
would choose the first versus the last redirecting target.  This is =
typically not a "configuration" as described in your example below =
(although, it is certainly possible).
>=20
> Typically, a voicemail system will choose the first one IF it is =
responsible for the mailbox of that target. Obviously, it would be =
impossible otherwise. In this latter case, the voicemail system would =
then use the last redirecting number.
>=20
> This is why I don't think we can put "normative" text, but instead =
"use case examples" as described above. A implementation could decide to =
always use the last redirecting number.
>=20
> On Nov 30, 2009, at 17:39 , Shida Schubert wrote:
>=20
>>=20
>> Francois;
>>=20
>> This was something Cullen asked we add to the draft too.
>>=20
>> I am assuming what we need is to add an explicit text that aids
>> voicemail server to identify the target of the voicemail box.
>>=20
>> If a call was initially received at Alice then diverted
>> to Carol,  then to Bob which then diverts to voicemail,
>> the question is for which AoR should the voicemail server
>> leave a voicemail?
>>=20
>> I think the answer depends on the configuration
>> of the voicemail server. But with the "mp" tag and
>> "mp" index we can identify when the mapping
>> (diversion) took place making it much more
>> deterministic to identify the target of the voicemail box.
>>=20
>> Looking at the example from the draft below;
>>=20
>> History-Info: <sip:bob@example.com>;index=3D1
>> History-Info: <sip:bob@192.0.2.3?Reason=3DSIP;cause=3D302>;\
>>                  index=3D1.1;rc
>>    History-Info: <sip:carol@example.com>;index=3D1.2;mp=3D1
>>    History-Info: <sip:carol@192.0.2.4>;index=3D1.2.1;rc
>>    History-Info: <sip:vm@example.com>;index=3D1.3;mp=3D1.2
>>    History-Info: <sip:vm@192.0.2.5>;index=3D1.3.1
>>=20
>> If the voicemail server is configured to use the voicemail box
>> of the initial target it can look for the first mp tag and look
>> at the hi-entry that mp index references (bob@example.com).
>>=20
>> If the voicemail server is configured to use the voicemail box
>> of the last target then it can look for the last mp tag and look
>> at the hi-entry that mp index references (carol@example.com) etc.
>>=20
>> Regards
>> Shida
>>=20
>> On Dec 1, 2009, at 7:56 AM, Francois AUDET wrote:
>>=20
>>>=20
>>> On Nov 30, 2009, at 5:13 , Alan Johnston wrote:
>>>=20
>>>>=20
>>>> Does this document have a good voicemail example?  This is the =
application I am most interested in, so it would be nice to have a very =
clear example of this.
>>>>=20
>>>=20
>>> That was the intent of section B.2 (in the appendix).
>>>=20
>>> Are you saying you'd want a better voicemail example? Or that we add =
more voicemail examples? What do you have in mind?
>>> _______________________________________________
>>> sipcore mailing list
>>> sipcore@ietf.org
>>> https://www.ietf.org/mailman/listinfo/sipcore
>>=20
>=20


From audet.francois@gmail.com  Tue Dec  1 21:55:58 2009
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From: Francois Audet <audet.francois@gmail.com>
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To: Shida Schubert <shida@ntt-at.com>, JENNINGS Cullen <fluffy@cisco.com>
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Cc: sipcore@ietf.org
Subject: Re: [sipcore] Some comments on draft-barnes-sipcore-rfc4244bis-03
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Cullen, Alan, what do you think?

On Dec 1, 2009, at 21:28 , Shida Schubert wrote:

>=20
> I don't think we need normative text, the whole examples are=20
> in appendix anyways, but I think what people want is to have=20
> explicit text on how one can determine the first versus the=20
> last redirecting target. I am not the one asking for them so=20
> I can't speak for them though..=20
>=20
> Regards
>  Shida
>=20
> On Dec 2, 2009, at 12:55 PM, Francois Audet wrote:
>=20
>> That seems pretty reasonable to me. But...
>>=20
>> Perhaps we can clarify a bit more in explaining why a voicemail =
system would choose the first versus the last redirecting target.  This =
is typically not a "configuration" as described in your example below =
(although, it is certainly possible).
>>=20
>> Typically, a voicemail system will choose the first one IF it is =
responsible for the mailbox of that target. Obviously, it would be =
impossible otherwise. In this latter case, the voicemail system would =
then use the last redirecting number.
>>=20
>> This is why I don't think we can put "normative" text, but instead =
"use case examples" as described above. A implementation could decide to =
always use the last redirecting number.
>>=20
>> On Nov 30, 2009, at 17:39 , Shida Schubert wrote:
>>=20
>>>=20
>>> Francois;
>>>=20
>>> This was something Cullen asked we add to the draft too.
>>>=20
>>> I am assuming what we need is to add an explicit text that aids
>>> voicemail server to identify the target of the voicemail box.
>>>=20
>>> If a call was initially received at Alice then diverted
>>> to Carol,  then to Bob which then diverts to voicemail,
>>> the question is for which AoR should the voicemail server
>>> leave a voicemail?
>>>=20
>>> I think the answer depends on the configuration
>>> of the voicemail server. But with the "mp" tag and
>>> "mp" index we can identify when the mapping
>>> (diversion) took place making it much more
>>> deterministic to identify the target of the voicemail box.
>>>=20
>>> Looking at the example from the draft below;
>>>=20
>>> History-Info: <sip:bob@example.com>;index=3D1
>>> History-Info: <sip:bob@192.0.2.3?Reason=3DSIP;cause=3D302>;\
>>>                 index=3D1.1;rc
>>>   History-Info: <sip:carol@example.com>;index=3D1.2;mp=3D1
>>>   History-Info: <sip:carol@192.0.2.4>;index=3D1.2.1;rc
>>>   History-Info: <sip:vm@example.com>;index=3D1.3;mp=3D1.2
>>>   History-Info: <sip:vm@192.0.2.5>;index=3D1.3.1
>>>=20
>>> If the voicemail server is configured to use the voicemail box
>>> of the initial target it can look for the first mp tag and look
>>> at the hi-entry that mp index references (bob@example.com).
>>>=20
>>> If the voicemail server is configured to use the voicemail box
>>> of the last target then it can look for the last mp tag and look
>>> at the hi-entry that mp index references (carol@example.com) etc.
>>>=20
>>> Regards
>>> Shida
>>>=20
>>> On Dec 1, 2009, at 7:56 AM, Francois AUDET wrote:
>>>=20
>>>>=20
>>>> On Nov 30, 2009, at 5:13 , Alan Johnston wrote:
>>>>=20
>>>>>=20
>>>>> Does this document have a good voicemail example?  This is the =
application I am most interested in, so it would be nice to have a very =
clear example of this.
>>>>>=20
>>>>=20
>>>> That was the intent of section B.2 (in the appendix).
>>>>=20
>>>> Are you saying you'd want a better voicemail example? Or that we =
add more voicemail examples? What do you have in mind?
>>>> _______________________________________________
>>>> sipcore mailing list
>>>> sipcore@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>=20
>>=20
>=20


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Thread-Topic: [sipcore] Some comments on draft-barnes-sipcore-rfc4244bis-03
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From: <R.Jesske@telekom.de>
To: <shida@ntt-at.com>, <audet.francois@gmail.com>
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Subject: Re: [sipcore] Some comments on draft-barnes-sipcore-rfc4244bis-03
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Hi Shida and Fran=E7ois,
I have never heard about an implementation that a forwarding service =
will choose the first entry for voicemail.
Should I see this more in the surrounding of private networks/PBX?

Seen from the diverting user either I would like to forward my call to =
an other URI or to an voicemail service. But that is a decision that the =
diverting user is taking.


Best Regards
Roland=20

-----Urspr=FCngliche Nachricht-----
Von: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] Im =
Auftrag von Shida Schubert
Gesendet: Mittwoch, 2. Dezember 2009 06:28
An: Francois Audet
Cc: sipcore@ietf.org
Betreff: Re: [sipcore] Some comments on =
draft-barnes-sipcore-rfc4244bis-03


 I don't think we need normative text, the whole examples are=20
in appendix anyways, but I think what people want is to have=20
explicit text on how one can determine the first versus the=20
last redirecting target. I am not the one asking for them so=20
I can't speak for them though..=20

 Regards
  Shida

On Dec 2, 2009, at 12:55 PM, Francois Audet wrote:

> That seems pretty reasonable to me. But...
>=20
> Perhaps we can clarify a bit more in explaining why a voicemail system =
would choose the first versus the last redirecting target.  This is =
typically not a "configuration" as described in your example below =
(although, it is certainly possible).
>=20
> Typically, a voicemail system will choose the first one IF it is =
responsible for the mailbox of that target. Obviously, it would be =
impossible otherwise. In this latter case, the voicemail system would =
then use the last redirecting number.
>=20
> This is why I don't think we can put "normative" text, but instead =
"use case examples" as described above. A implementation could decide to =
always use the last redirecting number.
>=20
> On Nov 30, 2009, at 17:39 , Shida Schubert wrote:
>=20
>>=20
>> Francois;
>>=20
>> This was something Cullen asked we add to the draft too.
>>=20
>> I am assuming what we need is to add an explicit text that aids
>> voicemail server to identify the target of the voicemail box.
>>=20
>> If a call was initially received at Alice then diverted
>> to Carol,  then to Bob which then diverts to voicemail,
>> the question is for which AoR should the voicemail server
>> leave a voicemail?
>>=20
>> I think the answer depends on the configuration
>> of the voicemail server. But with the "mp" tag and
>> "mp" index we can identify when the mapping
>> (diversion) took place making it much more
>> deterministic to identify the target of the voicemail box.
>>=20
>> Looking at the example from the draft below;
>>=20
>> History-Info: <sip:bob@example.com>;index=3D1
>> History-Info: <sip:bob@192.0.2.3?Reason=3DSIP;cause=3D302>;\
>>                  index=3D1.1;rc
>>    History-Info: <sip:carol@example.com>;index=3D1.2;mp=3D1
>>    History-Info: <sip:carol@192.0.2.4>;index=3D1.2.1;rc
>>    History-Info: <sip:vm@example.com>;index=3D1.3;mp=3D1.2
>>    History-Info: <sip:vm@192.0.2.5>;index=3D1.3.1
>>=20
>> If the voicemail server is configured to use the voicemail box
>> of the initial target it can look for the first mp tag and look
>> at the hi-entry that mp index references (bob@example.com).
>>=20
>> If the voicemail server is configured to use the voicemail box
>> of the last target then it can look for the last mp tag and look
>> at the hi-entry that mp index references (carol@example.com) etc.
>>=20
>> Regards
>> Shida
>>=20
>> On Dec 1, 2009, at 7:56 AM, Francois AUDET wrote:
>>=20
>>>=20
>>> On Nov 30, 2009, at 5:13 , Alan Johnston wrote:
>>>=20
>>>>=20
>>>> Does this document have a good voicemail example?  This is the =
application I am most interested in, so it would be nice to have a very =
clear example of this.
>>>>=20
>>>=20
>>> That was the intent of section B.2 (in the appendix).
>>>=20
>>> Are you saying you'd want a better voicemail example? Or that we add =
more voicemail examples? What do you have in mind?
>>> _______________________________________________
>>> sipcore mailing list
>>> sipcore@ietf.org
>>> https://www.ietf.org/mailman/listinfo/sipcore
>>=20
>=20

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

From shida@ntt-at.com  Wed Dec  2 00:47:38 2009
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From: Shida Schubert <shida@ntt-at.com>
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Cc: audet.francois@gmail.com, sipcore@ietf.org
Subject: Re: [sipcore] Some comments on draft-barnes-sipcore-rfc4244bis-03
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Roland;

 I agree for service provider like us it's always the=20
last diverting user that is the target for voicemail.=20

 I could imagine though that call may be diverted to=20
several people (AoR) before going to a voicemail server=20
within a corporate realm, in which case the target=20
may be the first entry who configured the server to=20
divert calls to several AoRs before diverting to a=20
voicemail server..=20

 Regards
  Shida

On Dec 2, 2009, at 5:12 PM, <R.Jesske@telekom.de> <R.Jesske@telekom.de> =
wrote:

> Hi Shida and Fran=E7ois,
> I have never heard about an implementation that a forwarding service =
will choose the first entry for voicemail.
> Should I see this more in the surrounding of private networks/PBX?
>=20
> Seen from the diverting user either I would like to forward my call to =
an other URI or to an voicemail service. But that is a decision that the =
diverting user is taking.
>=20
>=20
> Best Regards
> Roland=20
>=20
> -----Urspr=FCngliche Nachricht-----
> Von: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] Im =
Auftrag von Shida Schubert
> Gesendet: Mittwoch, 2. Dezember 2009 06:28
> An: Francois Audet
> Cc: sipcore@ietf.org
> Betreff: Re: [sipcore] Some comments on =
draft-barnes-sipcore-rfc4244bis-03
>=20
>=20
> I don't think we need normative text, the whole examples are=20
> in appendix anyways, but I think what people want is to have=20
> explicit text on how one can determine the first versus the=20
> last redirecting target. I am not the one asking for them so=20
> I can't speak for them though..=20
>=20
> Regards
>  Shida
>=20
> On Dec 2, 2009, at 12:55 PM, Francois Audet wrote:
>=20
>> That seems pretty reasonable to me. But...
>>=20
>> Perhaps we can clarify a bit more in explaining why a voicemail =
system would choose the first versus the last redirecting target.  This =
is typically not a "configuration" as described in your example below =
(although, it is certainly possible).
>>=20
>> Typically, a voicemail system will choose the first one IF it is =
responsible for the mailbox of that target. Obviously, it would be =
impossible otherwise. In this latter case, the voicemail system would =
then use the last redirecting number.
>>=20
>> This is why I don't think we can put "normative" text, but instead =
"use case examples" as described above. A implementation could decide to =
always use the last redirecting number.
>>=20
>> On Nov 30, 2009, at 17:39 , Shida Schubert wrote:
>>=20
>>>=20
>>> Francois;
>>>=20
>>> This was something Cullen asked we add to the draft too.
>>>=20
>>> I am assuming what we need is to add an explicit text that aids
>>> voicemail server to identify the target of the voicemail box.
>>>=20
>>> If a call was initially received at Alice then diverted
>>> to Carol,  then to Bob which then diverts to voicemail,
>>> the question is for which AoR should the voicemail server
>>> leave a voicemail?
>>>=20
>>> I think the answer depends on the configuration
>>> of the voicemail server. But with the "mp" tag and
>>> "mp" index we can identify when the mapping
>>> (diversion) took place making it much more
>>> deterministic to identify the target of the voicemail box.
>>>=20
>>> Looking at the example from the draft below;
>>>=20
>>> History-Info: <sip:bob@example.com>;index=3D1
>>> History-Info: <sip:bob@192.0.2.3?Reason=3DSIP;cause=3D302>;\
>>>                 index=3D1.1;rc
>>>   History-Info: <sip:carol@example.com>;index=3D1.2;mp=3D1
>>>   History-Info: <sip:carol@192.0.2.4>;index=3D1.2.1;rc
>>>   History-Info: <sip:vm@example.com>;index=3D1.3;mp=3D1.2
>>>   History-Info: <sip:vm@192.0.2.5>;index=3D1.3.1
>>>=20
>>> If the voicemail server is configured to use the voicemail box
>>> of the initial target it can look for the first mp tag and look
>>> at the hi-entry that mp index references (bob@example.com).
>>>=20
>>> If the voicemail server is configured to use the voicemail box
>>> of the last target then it can look for the last mp tag and look
>>> at the hi-entry that mp index references (carol@example.com) etc.
>>>=20
>>> Regards
>>> Shida
>>>=20
>>> On Dec 1, 2009, at 7:56 AM, Francois AUDET wrote:
>>>=20
>>>>=20
>>>> On Nov 30, 2009, at 5:13 , Alan Johnston wrote:
>>>>=20
>>>>>=20
>>>>> Does this document have a good voicemail example?  This is the =
application I am most interested in, so it would be nice to have a very =
clear example of this.
>>>>>=20
>>>>=20
>>>> That was the intent of section B.2 (in the appendix).
>>>>=20
>>>> Are you saying you'd want a better voicemail example? Or that we =
add more voicemail examples? What do you have in mind?
>>>> _______________________________________________
>>>> sipcore mailing list
>>>> sipcore@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>=20
>>=20
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From christer.holmberg@ericsson.com  Wed Dec  2 00:59:41 2009
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Thread-Topic: Draft new version: draft-ietf-sipcore-info-events
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
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Subject: [sipcore] Draft new version: draft-ietf-sipcore-info-events
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Hi,

I've submitted a new version of the info draft. Hopefully all comments
and feedback have now been addressed.

The biggest changes are:

- Section 4.2: The usage of the Recv-Info header has been clarified,
some simple rules have been added, and there were some discussions that
since the re-INVITE fallback issue also affects Recv-Info (if you
receive a Recv-Info in a 18x to a re-INVITE, and the re-INVITE then
fails) and that we need some words about that. The text now says that IF
the request contains a Recv-Info header, the response must also contain
it.

- Section 11.4: Contains new wording about the IANA process for
registering info packages, including the expert review etc.

- Section 12: Additional examples have been added


Please note that I did NOT remove Recv-Info in PRACKs, since there in my
opinion was no concensus to do so.


The draft can also be found at:

http://users.piuha.net/cholmber/drafts/draft-ietf-sipcore-info-events-03
.txt


Regards,

Christer

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charset=3Dus-ascii">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
6.5.7654.12">
<TITLE>Draft new version: draft-ietf-sipcore-info-events</TITLE>
</HEAD>
<BODY>
<!-- Converted from text/rtf format -->
<BR>

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

<P><FONT SIZE=3D2 FACE=3D"Arial">I've submitted a new version of the =
info draft. Hopefully all comments and feedback have now been =
addressed.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">The biggest changes are:</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">- Section 4.2: The usage of the =
Recv-Info header has been clarified, some simple rules have been added, =
and there were some discussions that since the re-INVITE fallback issue =
also affects Recv-Info (if you receive a Recv-Info in a 18x to a =
re-INVITE, and the re-INVITE then fails) and that we need some words =
about that. The text now says that IF the request contains a Recv-Info =
header, the response must also contain it.</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">- Section 11.4: Contains new wording =
about the IANA process for registering info packages, including the =
expert review etc.</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">- Section 12: Additional examples have =
been added</FONT>
</P>
<BR>

<P><FONT SIZE=3D2 FACE=3D"Arial">Please note that I did NOT remove =
Recv-Info in PRACKs, since there in my opinion was no concensus to do =
so.</FONT>
</P>
<BR>

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

<P><A =
HREF=3D"http://users.piuha.net/cholmber/drafts/draft-ietf-sipcore-info-ev=
ents-03.txt"><U><FONT COLOR=3D"#0000FF" SIZE=3D2 =
FACE=3D"Arial">http://users.piuha.net/cholmber/drafts/draft-ietf-sipcore-=
info-events-03.txt</FONT></U></A>
</P>
<BR>

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

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

</BODY>
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	Title           : Session Initiation Protocol (SIP) INFO Method and Package Framework
	Author(s)       : E. Burger, et al.
	Filename        : draft-ietf-sipcore-info-events-03.txt
	Pages           : 38
	Date            : 2009-12-02

This document defines a new method, INFO, for the Session Initiation
Protocol (SIP) [RFC3261], and an Info Package mechanism.  The
document obsoletes [RFC2976].  For backward compatibility the
document also specifies a "legacy" mode of usage of the INFO method
that is compatible with the usage previously defined in [RFC2976],
referred to as "legacy INFO Usage" in this document.

Conventions Used in this Document

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document are to be interpreted as described in RFC 2119 [RFC2119].
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Hi,

I've posted a new version (-01) of the 199 draft, based on Robert's AD
comments.

Regards,

Christer

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This draft is a work item of the Session Initiation Protocol Core Working Group of the IETF.


	Title           : Response Code for Indication of Terminated Dialog
	Author(s)       : C. Holmberg
	Filename        : draft-ietf-sipcore-199-01.txt
	Pages           : 16
	Date            : 2009-12-02

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

Status of this Memo

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Cc: audet.francois@gmail.com, sipcore@ietf.org
Subject: Re: [sipcore] Some comments on draft-barnes-sipcore-rfc4244bis-03
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R.Jesske@telekom.de wrote:
> Hi Shida and François,
> I have never heard about an implementation that a forwarding service will choose the first entry for voicemail.
> Should I see this more in the surrounding of private networks/PBX?
> 
> Seen from the diverting user either I would like to forward my call to an other URI or to an voicemail service. But that is a decision that the diverting user is taking.

IMO the diverting user should have the option to attempt a diversion to 
another number but have the call go to his own voicemail if that 
diversion fails - preempting subsequent diversion to another's voicemail.

But that should *not* depend on on the downstream users using the same 
VM server. Its a matter of the diverting user entering a preference that 
the call not go to voicemail downstream. Callerprefs has a mechanism to 
indicate this, but I have never heard of it being supported for this.

This would of course be *policy*. In other cases I might indeed want to 
delegate full responsibility for the call, including going to voicemail.

Unfortunately, making this work requires that the diversion target 
cooperate.

	Thanks,
	Paul

> 
> Best Regards
> Roland 
> 
> -----Ursprüngliche Nachricht-----
> Von: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] Im Auftrag von Shida Schubert
> Gesendet: Mittwoch, 2. Dezember 2009 06:28
> An: Francois Audet
> Cc: sipcore@ietf.org
> Betreff: Re: [sipcore] Some comments on draft-barnes-sipcore-rfc4244bis-03
> 
> 
>  I don't think we need normative text, the whole examples are 
> in appendix anyways, but I think what people want is to have 
> explicit text on how one can determine the first versus the 
> last redirecting target. I am not the one asking for them so 
> I can't speak for them though.. 
> 
>  Regards
>   Shida
> 
> On Dec 2, 2009, at 12:55 PM, Francois Audet wrote:
> 
>> That seems pretty reasonable to me. But...
>>
>> Perhaps we can clarify a bit more in explaining why a voicemail system would choose the first versus the last redirecting target.  This is typically not a "configuration" as described in your example below (although, it is certainly possible).
>>
>> Typically, a voicemail system will choose the first one IF it is responsible for the mailbox of that target. Obviously, it would be impossible otherwise. In this latter case, the voicemail system would then use the last redirecting number.
>>
>> This is why I don't think we can put "normative" text, but instead "use case examples" as described above. A implementation could decide to always use the last redirecting number.
>>
>> On Nov 30, 2009, at 17:39 , Shida Schubert wrote:
>>
>>> Francois;
>>>
>>> This was something Cullen asked we add to the draft too.
>>>
>>> I am assuming what we need is to add an explicit text that aids
>>> voicemail server to identify the target of the voicemail box.
>>>
>>> If a call was initially received at Alice then diverted
>>> to Carol,  then to Bob which then diverts to voicemail,
>>> the question is for which AoR should the voicemail server
>>> leave a voicemail?
>>>
>>> I think the answer depends on the configuration
>>> of the voicemail server. But with the "mp" tag and
>>> "mp" index we can identify when the mapping
>>> (diversion) took place making it much more
>>> deterministic to identify the target of the voicemail box.
>>>
>>> Looking at the example from the draft below;
>>>
>>> History-Info: <sip:bob@example.com>;index=1
>>> History-Info: <sip:bob@192.0.2.3?Reason=SIP;cause=302>;\
>>>                  index=1.1;rc
>>>    History-Info: <sip:carol@example.com>;index=1.2;mp=1
>>>    History-Info: <sip:carol@192.0.2.4>;index=1.2.1;rc
>>>    History-Info: <sip:vm@example.com>;index=1.3;mp=1.2
>>>    History-Info: <sip:vm@192.0.2.5>;index=1.3.1
>>>
>>> If the voicemail server is configured to use the voicemail box
>>> of the initial target it can look for the first mp tag and look
>>> at the hi-entry that mp index references (bob@example.com).
>>>
>>> If the voicemail server is configured to use the voicemail box
>>> of the last target then it can look for the last mp tag and look
>>> at the hi-entry that mp index references (carol@example.com) etc.
>>>
>>> Regards
>>> Shida
>>>
>>> On Dec 1, 2009, at 7:56 AM, Francois AUDET wrote:
>>>
>>>> On Nov 30, 2009, at 5:13 , Alan Johnston wrote:
>>>>
>>>>> Does this document have a good voicemail example?  This is the application I am most interested in, so it would be nice to have a very clear example of this.
>>>>>
>>>> That was the intent of section B.2 (in the appendix).
>>>>
>>>> Are you saying you'd want a better voicemail example? Or that we add more voicemail examples? What do you have in mind?
>>>> _______________________________________________
>>>> 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
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> 

From R.Jesske@telekom.de  Wed Dec  2 07:07:09 2009
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From: <R.Jesske@telekom.de>
To: <pkyzivat@cisco.com>
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Cc: audet.francois@gmail.com, sipcore@ietf.org
Subject: Re: [sipcore] Some comments on draft-barnes-sipcore-rfc4244bis-03
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The only application I could think about is a company's network where =
one call agent has a couple of replacements and if they are not =
reachable then the voicemail of the originally called UA is then put in =
charge.

But As Paul said that is a policy within a private network/PBX and will =
never happen for an operator approach. So if such an example will be put =
into, then the circumstances should be described explicitly.

Thanks you and Best Regards

Roland=20

-----Urspr=FCngliche Nachricht-----
Von: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
Gesendet: Mittwoch, 2. Dezember 2009 15:01
An: Jesske, Roland
Cc: shida@ntt-at.com; audet.francois@gmail.com; sipcore@ietf.org
Betreff: Re: [sipcore] Some comments on =
draft-barnes-sipcore-rfc4244bis-03



R.Jesske@telekom.de wrote:
> Hi Shida and Fran=E7ois,
> I have never heard about an implementation that a forwarding service =
will choose the first entry for voicemail.
> Should I see this more in the surrounding of private networks/PBX?
>=20
> Seen from the diverting user either I would like to forward my call to =
an other URI or to an voicemail service. But that is a decision that the =
diverting user is taking.

IMO the diverting user should have the option to attempt a diversion to=20
another number but have the call go to his own voicemail if that=20
diversion fails - preempting subsequent diversion to another's =
voicemail.

But that should *not* depend on on the downstream users using the same=20
VM server. Its a matter of the diverting user entering a preference that =

the call not go to voicemail downstream. Callerprefs has a mechanism to=20
indicate this, but I have never heard of it being supported for this.

This would of course be *policy*. In other cases I might indeed want to=20
delegate full responsibility for the call, including going to voicemail.

Unfortunately, making this work requires that the diversion target=20
cooperate.

	Thanks,
	Paul

>=20
> Best Regards
> Roland=20
>=20
> -----Urspr=FCngliche Nachricht-----
> Von: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] Im =
Auftrag von Shida Schubert
> Gesendet: Mittwoch, 2. Dezember 2009 06:28
> An: Francois Audet
> Cc: sipcore@ietf.org
> Betreff: Re: [sipcore] Some comments on =
draft-barnes-sipcore-rfc4244bis-03
>=20
>=20
>  I don't think we need normative text, the whole examples are=20
> in appendix anyways, but I think what people want is to have=20
> explicit text on how one can determine the first versus the=20
> last redirecting target. I am not the one asking for them so=20
> I can't speak for them though..=20
>=20
>  Regards
>   Shida
>=20
> On Dec 2, 2009, at 12:55 PM, Francois Audet wrote:
>=20
>> That seems pretty reasonable to me. But...
>>
>> Perhaps we can clarify a bit more in explaining why a voicemail =
system would choose the first versus the last redirecting target.  This =
is typically not a "configuration" as described in your example below =
(although, it is certainly possible).
>>
>> Typically, a voicemail system will choose the first one IF it is =
responsible for the mailbox of that target. Obviously, it would be =
impossible otherwise. In this latter case, the voicemail system would =
then use the last redirecting number.
>>
>> This is why I don't think we can put "normative" text, but instead =
"use case examples" as described above. A implementation could decide to =
always use the last redirecting number.
>>
>> On Nov 30, 2009, at 17:39 , Shida Schubert wrote:
>>
>>> Francois;
>>>
>>> This was something Cullen asked we add to the draft too.
>>>
>>> I am assuming what we need is to add an explicit text that aids
>>> voicemail server to identify the target of the voicemail box.
>>>
>>> If a call was initially received at Alice then diverted
>>> to Carol,  then to Bob which then diverts to voicemail,
>>> the question is for which AoR should the voicemail server
>>> leave a voicemail?
>>>
>>> I think the answer depends on the configuration
>>> of the voicemail server. But with the "mp" tag and
>>> "mp" index we can identify when the mapping
>>> (diversion) took place making it much more
>>> deterministic to identify the target of the voicemail box.
>>>
>>> Looking at the example from the draft below;
>>>
>>> History-Info: <sip:bob@example.com>;index=3D1
>>> History-Info: <sip:bob@192.0.2.3?Reason=3DSIP;cause=3D302>;\
>>>                  index=3D1.1;rc
>>>    History-Info: <sip:carol@example.com>;index=3D1.2;mp=3D1
>>>    History-Info: <sip:carol@192.0.2.4>;index=3D1.2.1;rc
>>>    History-Info: <sip:vm@example.com>;index=3D1.3;mp=3D1.2
>>>    History-Info: <sip:vm@192.0.2.5>;index=3D1.3.1
>>>
>>> If the voicemail server is configured to use the voicemail box
>>> of the initial target it can look for the first mp tag and look
>>> at the hi-entry that mp index references (bob@example.com).
>>>
>>> If the voicemail server is configured to use the voicemail box
>>> of the last target then it can look for the last mp tag and look
>>> at the hi-entry that mp index references (carol@example.com) etc.
>>>
>>> Regards
>>> Shida
>>>
>>> On Dec 1, 2009, at 7:56 AM, Francois AUDET wrote:
>>>
>>>> On Nov 30, 2009, at 5:13 , Alan Johnston wrote:
>>>>
>>>>> Does this document have a good voicemail example?  This is the =
application I am most interested in, so it would be nice to have a very =
clear example of this.
>>>>>
>>>> That was the intent of section B.2 (in the appendix).
>>>>
>>>> Are you saying you'd want a better voicemail example? Or that we =
add more voicemail examples? What do you have in mind?
>>>> _______________________________________________
>>>> sipcore mailing list
>>>> sipcore@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/sipcore
>=20
> _______________________________________________
> 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
>=20

From audet.francois@gmail.com  Wed Dec  2 09:20:18 2009
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From: Francois Audet <audet.francois@gmail.com>
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Date: Wed, 2 Dec 2009 09:19:58 -0800
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References: <4B13C4FC.5050907@sipstation.com><98E67934-70CE-49E4-A2C3-05C3EB42573F@skype.net><C25ED9E1-DA40-4D79-92FA-148958AE2154@ntt-at.com><10A79C23-41FB-4FF8-AD55-2F83D7017EFC@gmail.com> <057B55FD-2B09-4296-AB60-6FA7E47EDD37@ntt-at.com> <9886E5FCA6D76549A3011068483A4BD40551C3B6@S4DE8PSAAQB.mitte.t-com.de>
To: <R.Jesske@telekom.de>
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Cc: sipcore@ietf.org
Subject: Re: [sipcore] Some comments on draft-barnes-sipcore-rfc4244bis-03
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Yeah, PBXs do this.

I was told that service providers do this to.

When you think about it, it makes sense.

If you call me, you would expect to get my voicemail, not somebody =
else's.

On Dec 2, 2009, at 0:12 , <R.Jesske@telekom.de> wrote:

> Hi Shida and Fran=E7ois,
> I have never heard about an implementation that a forwarding service =
will choose the first entry for voicemail.
> Should I see this more in the surrounding of private networks/PBX?
>=20
> Seen from the diverting user either I would like to forward my call to =
an other URI or to an voicemail service. But that is a decision that the =
diverting user is taking.
>=20
>=20
> Best Regards
> Roland=20
>=20
> -----Urspr=FCngliche Nachricht-----
> Von: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] Im =
Auftrag von Shida Schubert
> Gesendet: Mittwoch, 2. Dezember 2009 06:28
> An: Francois Audet
> Cc: sipcore@ietf.org
> Betreff: Re: [sipcore] Some comments on =
draft-barnes-sipcore-rfc4244bis-03
>=20
>=20
> I don't think we need normative text, the whole examples are=20
> in appendix anyways, but I think what people want is to have=20
> explicit text on how one can determine the first versus the=20
> last redirecting target. I am not the one asking for them so=20
> I can't speak for them though..=20
>=20
> Regards
>  Shida
>=20
> On Dec 2, 2009, at 12:55 PM, Francois Audet wrote:
>=20
>> That seems pretty reasonable to me. But...
>>=20
>> Perhaps we can clarify a bit more in explaining why a voicemail =
system would choose the first versus the last redirecting target.  This =
is typically not a "configuration" as described in your example below =
(although, it is certainly possible).
>>=20
>> Typically, a voicemail system will choose the first one IF it is =
responsible for the mailbox of that target. Obviously, it would be =
impossible otherwise. In this latter case, the voicemail system would =
then use the last redirecting number.
>>=20
>> This is why I don't think we can put "normative" text, but instead =
"use case examples" as described above. A implementation could decide to =
always use the last redirecting number.
>>=20
>> On Nov 30, 2009, at 17:39 , Shida Schubert wrote:
>>=20
>>>=20
>>> Francois;
>>>=20
>>> This was something Cullen asked we add to the draft too.
>>>=20
>>> I am assuming what we need is to add an explicit text that aids
>>> voicemail server to identify the target of the voicemail box.
>>>=20
>>> If a call was initially received at Alice then diverted
>>> to Carol,  then to Bob which then diverts to voicemail,
>>> the question is for which AoR should the voicemail server
>>> leave a voicemail?
>>>=20
>>> I think the answer depends on the configuration
>>> of the voicemail server. But with the "mp" tag and
>>> "mp" index we can identify when the mapping
>>> (diversion) took place making it much more
>>> deterministic to identify the target of the voicemail box.
>>>=20
>>> Looking at the example from the draft below;
>>>=20
>>> History-Info: <sip:bob@example.com>;index=3D1
>>> History-Info: <sip:bob@192.0.2.3?Reason=3DSIP;cause=3D302>;\
>>>                 index=3D1.1;rc
>>>   History-Info: <sip:carol@example.com>;index=3D1.2;mp=3D1
>>>   History-Info: <sip:carol@192.0.2.4>;index=3D1.2.1;rc
>>>   History-Info: <sip:vm@example.com>;index=3D1.3;mp=3D1.2
>>>   History-Info: <sip:vm@192.0.2.5>;index=3D1.3.1
>>>=20
>>> If the voicemail server is configured to use the voicemail box
>>> of the initial target it can look for the first mp tag and look
>>> at the hi-entry that mp index references (bob@example.com).
>>>=20
>>> If the voicemail server is configured to use the voicemail box
>>> of the last target then it can look for the last mp tag and look
>>> at the hi-entry that mp index references (carol@example.com) etc.
>>>=20
>>> Regards
>>> Shida
>>>=20
>>> On Dec 1, 2009, at 7:56 AM, Francois AUDET wrote:
>>>=20
>>>>=20
>>>> On Nov 30, 2009, at 5:13 , Alan Johnston wrote:
>>>>=20
>>>>>=20
>>>>> Does this document have a good voicemail example?  This is the =
application I am most interested in, so it would be nice to have a very =
clear example of this.
>>>>>=20
>>>>=20
>>>> That was the intent of section B.2 (in the appendix).
>>>>=20
>>>> Are you saying you'd want a better voicemail example? Or that we =
add more voicemail examples? What do you have in mind?
>>>> _______________________________________________
>>>> sipcore mailing list
>>>> sipcore@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>=20
>>=20
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From pkyzivat@cisco.com  Wed Dec  2 09:25:36 2009
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Cc: audet.francois@gmail.com, sipcore@ietf.org
Subject: Re: [sipcore] Some comments on draft-barnes-sipcore-rfc4244bis-03
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R.Jesske@telekom.de wrote:
> The only application I could think about is a company's network where one call agent has a couple of replacements and if they are not reachable then the voicemail of the originally called UA is then put in charge.
> 
> But As Paul said that is a policy within a private network/PBX and will never happen for an operator approach. So if such an example will be put into, then the circumstances should be described explicitly.

Some cases that are (IMO) very plausible:

1) I want incoming calls to my cellphone to divert to my home phone
    if I don't answer. But if there is no answer at home either,
    then I want the call to go to my cellphone voicemail. (Because
    I'm more likely to check that in a timely way.)

2) I want calls to my office phone to try my personal cell phone
    if I'm not in the office. But I want any messages left on my
    office voicemail because it is corporate policy not to have
    business stuff recorded on VM servers it doesn't control.

3) When I'm at lunch, I want calls to my business line to try
    the person who backs me up when I'm away. But if that person
    can't take the call then I want it on my voicemail to deal
    with after lunch.

    (Converse to this: When I'm on *vacation*, I want calls to my
    business line to try the person who backs me up. If they can't
    answer, then I want messages left on *their* vm, so that they
    can deal with it later. Don't want callers waiting until I get
    back from vacation.)

	Thanks,
	Paul

> Thanks you and Best Regards
> 
> Roland 
> 
> -----Ursprüngliche Nachricht-----
> Von: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
> Gesendet: Mittwoch, 2. Dezember 2009 15:01
> An: Jesske, Roland
> Cc: shida@ntt-at.com; audet.francois@gmail.com; sipcore@ietf.org
> Betreff: Re: [sipcore] Some comments on draft-barnes-sipcore-rfc4244bis-03
> 
> 
> 
> R.Jesske@telekom.de wrote:
>> Hi Shida and François,
>> I have never heard about an implementation that a forwarding service will choose the first entry for voicemail.
>> Should I see this more in the surrounding of private networks/PBX?
>>
>> Seen from the diverting user either I would like to forward my call to an other URI or to an voicemail service. But that is a decision that the diverting user is taking.
> 
> IMO the diverting user should have the option to attempt a diversion to 
> another number but have the call go to his own voicemail if that 
> diversion fails - preempting subsequent diversion to another's voicemail.
> 
> But that should *not* depend on on the downstream users using the same 
> VM server. Its a matter of the diverting user entering a preference that 
> the call not go to voicemail downstream. Callerprefs has a mechanism to 
> indicate this, but I have never heard of it being supported for this.
> 
> This would of course be *policy*. In other cases I might indeed want to 
> delegate full responsibility for the call, including going to voicemail.
> 
> Unfortunately, making this work requires that the diversion target 
> cooperate.
> 
> 	Thanks,
> 	Paul
> 
>> Best Regards
>> Roland 
>>
>> -----Ursprüngliche Nachricht-----
>> Von: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] Im Auftrag von Shida Schubert
>> Gesendet: Mittwoch, 2. Dezember 2009 06:28
>> An: Francois Audet
>> Cc: sipcore@ietf.org
>> Betreff: Re: [sipcore] Some comments on draft-barnes-sipcore-rfc4244bis-03
>>
>>
>>  I don't think we need normative text, the whole examples are 
>> in appendix anyways, but I think what people want is to have 
>> explicit text on how one can determine the first versus the 
>> last redirecting target. I am not the one asking for them so 
>> I can't speak for them though.. 
>>
>>  Regards
>>   Shida
>>
>> On Dec 2, 2009, at 12:55 PM, Francois Audet wrote:
>>
>>> That seems pretty reasonable to me. But...
>>>
>>> Perhaps we can clarify a bit more in explaining why a voicemail system would choose the first versus the last redirecting target.  This is typically not a "configuration" as described in your example below (although, it is certainly possible).
>>>
>>> Typically, a voicemail system will choose the first one IF it is responsible for the mailbox of that target. Obviously, it would be impossible otherwise. In this latter case, the voicemail system would then use the last redirecting number.
>>>
>>> This is why I don't think we can put "normative" text, but instead "use case examples" as described above. A implementation could decide to always use the last redirecting number.
>>>
>>> On Nov 30, 2009, at 17:39 , Shida Schubert wrote:
>>>
>>>> Francois;
>>>>
>>>> This was something Cullen asked we add to the draft too.
>>>>
>>>> I am assuming what we need is to add an explicit text that aids
>>>> voicemail server to identify the target of the voicemail box.
>>>>
>>>> If a call was initially received at Alice then diverted
>>>> to Carol,  then to Bob which then diverts to voicemail,
>>>> the question is for which AoR should the voicemail server
>>>> leave a voicemail?
>>>>
>>>> I think the answer depends on the configuration
>>>> of the voicemail server. But with the "mp" tag and
>>>> "mp" index we can identify when the mapping
>>>> (diversion) took place making it much more
>>>> deterministic to identify the target of the voicemail box.
>>>>
>>>> Looking at the example from the draft below;
>>>>
>>>> History-Info: <sip:bob@example.com>;index=1
>>>> History-Info: <sip:bob@192.0.2.3?Reason=SIP;cause=302>;\
>>>>                  index=1.1;rc
>>>>    History-Info: <sip:carol@example.com>;index=1.2;mp=1
>>>>    History-Info: <sip:carol@192.0.2.4>;index=1.2.1;rc
>>>>    History-Info: <sip:vm@example.com>;index=1.3;mp=1.2
>>>>    History-Info: <sip:vm@192.0.2.5>;index=1.3.1
>>>>
>>>> If the voicemail server is configured to use the voicemail box
>>>> of the initial target it can look for the first mp tag and look
>>>> at the hi-entry that mp index references (bob@example.com).
>>>>
>>>> If the voicemail server is configured to use the voicemail box
>>>> of the last target then it can look for the last mp tag and look
>>>> at the hi-entry that mp index references (carol@example.com) etc.
>>>>
>>>> Regards
>>>> Shida
>>>>
>>>> On Dec 1, 2009, at 7:56 AM, Francois AUDET wrote:
>>>>
>>>>> On Nov 30, 2009, at 5:13 , Alan Johnston wrote:
>>>>>
>>>>>> Does this document have a good voicemail example?  This is the application I am most interested in, so it would be nice to have a very clear example of this.
>>>>>>
>>>>> That was the intent of section B.2 (in the appendix).
>>>>>
>>>>> Are you saying you'd want a better voicemail example? Or that we add more voicemail examples? What do you have in mind?
>>>>> _______________________________________________
>>>>> 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
>>
> 

From john.elwell@siemens-enterprise.com  Wed Dec  2 12:15:49 2009
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Paul Kyzivat <pkyzivat@cisco.com>, "R.Jesske@telekom.de" <R.Jesske@telekom.de>
Date: Wed, 2 Dec 2009 21:15:34 +0100
Thread-Topic: [sipcore] Some comments on draft-barnes-sipcore-rfc4244bis-03
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References: <4B13C4FC.5050907@sipstation.com><98E67934-70CE-49E4-A2C3-05C3EB42573F@skype.net><C25ED9E1-DA40-4D79-92FA-148958AE2154@ntt-at.com><10A79C23-41FB-4FF8-AD55-2F83D7017EFC@gmail.com> <057B55FD-2B09-4296-AB60-6FA7E47EDD37@ntt-at.com> <9886E5FCA6D76549A3011068483A4BD40551C3B6@S4DE8PSAAQB.mitte.t-com.de> <4B16731E.20000@cisco.com> <9886E5FCA6D76549A3011068483A4BD40551C7A0@S4DE8PSAAQB.mitte.t-com.de> <4B16A305.5090700@cisco.com>
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Cc: "audet.francois@gmail.com" <audet.francois@gmail.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] Some comments on draft-barnes-sipcore-rfc4244bis-03
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I agree with these examples Paul has enumerated, and I personally use the s=
econd one. It relies on my domain proxy stepping in and cancelling the call=
 to my cellphone before it has chance to divert to cellphone (personal) voi=
cemail (unless of course I have disabled that capability). As Paul pointed =
out earlier, SIP does have a means of doing it using caller preferences, bu=
t this doesn't work if the call goes to PSTN.

John

> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Paul Kyzivat
> Sent: 02 December 2009 17:25
> To: R.Jesske@telekom.de
> Cc: audet.francois@gmail.com; sipcore@ietf.org
> Subject: Re: [sipcore] Some comments on=20
> draft-barnes-sipcore-rfc4244bis-03
>=20
>=20
>=20
> R.Jesske@telekom.de wrote:
> > The only application I could think about is a company's=20
> network where one call agent has a couple of replacements and=20
> if they are not reachable then the voicemail of the=20
> originally called UA is then put in charge.
> >=20
> > But As Paul said that is a policy within a private=20
> network/PBX and will never happen for an operator approach.=20
> So if such an example will be put into, then the=20
> circumstances should be described explicitly.
>=20
> Some cases that are (IMO) very plausible:
>=20
> 1) I want incoming calls to my cellphone to divert to my home phone
>     if I don't answer. But if there is no answer at home either,
>     then I want the call to go to my cellphone voicemail. (Because
>     I'm more likely to check that in a timely way.)
>=20
> 2) I want calls to my office phone to try my personal cell phone
>     if I'm not in the office. But I want any messages left on my
>     office voicemail because it is corporate policy not to have
>     business stuff recorded on VM servers it doesn't control.
>=20
> 3) When I'm at lunch, I want calls to my business line to try
>     the person who backs me up when I'm away. But if that person
>     can't take the call then I want it on my voicemail to deal
>     with after lunch.
>=20
>     (Converse to this: When I'm on *vacation*, I want calls to my
>     business line to try the person who backs me up. If they can't
>     answer, then I want messages left on *their* vm, so that they
>     can deal with it later. Don't want callers waiting until I get
>     back from vacation.)
>=20
> 	Thanks,
> 	Paul
>=20
> > Thanks you and Best Regards
> >=20
> > Roland=20
> >=20
> > -----Urspr=FCngliche Nachricht-----
> > Von: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
> > Gesendet: Mittwoch, 2. Dezember 2009 15:01
> > An: Jesske, Roland
> > Cc: shida@ntt-at.com; audet.francois@gmail.com; sipcore@ietf.org
> > Betreff: Re: [sipcore] Some comments on=20
> draft-barnes-sipcore-rfc4244bis-03
> >=20
> >=20
> >=20
> > R.Jesske@telekom.de wrote:
> >> Hi Shida and Fran=E7ois,
> >> I have never heard about an implementation that a=20
> forwarding service will choose the first entry for voicemail.
> >> Should I see this more in the surrounding of private networks/PBX?
> >>
> >> Seen from the diverting user either I would like to=20
> forward my call to an other URI or to an voicemail service.=20
> But that is a decision that the diverting user is taking.
> >=20
> > IMO the diverting user should have the option to attempt a=20
> diversion to=20
> > another number but have the call go to his own voicemail if that=20
> > diversion fails - preempting subsequent diversion to=20
> another's voicemail.
> >=20
> > But that should *not* depend on on the downstream users=20
> using the same=20
> > VM server. Its a matter of the diverting user entering a=20
> preference that=20
> > the call not go to voicemail downstream. Callerprefs has a=20
> mechanism to=20
> > indicate this, but I have never heard of it being supported=20
> for this.
> >=20
> > This would of course be *policy*. In other cases I might=20
> indeed want to=20
> > delegate full responsibility for the call, including going=20
> to voicemail.
> >=20
> > Unfortunately, making this work requires that the diversion target=20
> > cooperate.
> >=20
> > 	Thanks,
> > 	Paul
> >=20
> >> Best Regards
> >> Roland=20
> >>
> >> -----Urspr=FCngliche Nachricht-----
> >> Von: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] Im Auftrag von Shida Schubert
> >> Gesendet: Mittwoch, 2. Dezember 2009 06:28
> >> An: Francois Audet
> >> Cc: sipcore@ietf.org
> >> Betreff: Re: [sipcore] Some comments on=20
> draft-barnes-sipcore-rfc4244bis-03
> >>
> >>
> >>  I don't think we need normative text, the whole examples are=20
> >> in appendix anyways, but I think what people want is to have=20
> >> explicit text on how one can determine the first versus the=20
> >> last redirecting target. I am not the one asking for them so=20
> >> I can't speak for them though..=20
> >>
> >>  Regards
> >>   Shida
> >>
> >> On Dec 2, 2009, at 12:55 PM, Francois Audet wrote:
> >>
> >>> That seems pretty reasonable to me. But...
> >>>
> >>> Perhaps we can clarify a bit more in explaining why a=20
> voicemail system would choose the first versus the last=20
> redirecting target.  This is typically not a "configuration"=20
> as described in your example below (although, it is certainly=20
> possible).
> >>>
> >>> Typically, a voicemail system will choose the first one=20
> IF it is responsible for the mailbox of that target.=20
> Obviously, it would be impossible otherwise. In this latter=20
> case, the voicemail system would then use the last redirecting number.
> >>>
> >>> This is why I don't think we can put "normative" text,=20
> but instead "use case examples" as described above. A=20
> implementation could decide to always use the last redirecting number.
> >>>
> >>> On Nov 30, 2009, at 17:39 , Shida Schubert wrote:
> >>>
> >>>> Francois;
> >>>>
> >>>> This was something Cullen asked we add to the draft too.
> >>>>
> >>>> I am assuming what we need is to add an explicit text that aids
> >>>> voicemail server to identify the target of the voicemail box.
> >>>>
> >>>> If a call was initially received at Alice then diverted
> >>>> to Carol,  then to Bob which then diverts to voicemail,
> >>>> the question is for which AoR should the voicemail server
> >>>> leave a voicemail?
> >>>>
> >>>> I think the answer depends on the configuration
> >>>> of the voicemail server. But with the "mp" tag and
> >>>> "mp" index we can identify when the mapping
> >>>> (diversion) took place making it much more
> >>>> deterministic to identify the target of the voicemail box.
> >>>>
> >>>> Looking at the example from the draft below;
> >>>>
> >>>> History-Info: <sip:bob@example.com>;index=3D1
> >>>> History-Info: <sip:bob@192.0.2.3?Reason=3DSIP;cause=3D302>;\
> >>>>                  index=3D1.1;rc
> >>>>    History-Info: <sip:carol@example.com>;index=3D1.2;mp=3D1
> >>>>    History-Info: <sip:carol@192.0.2.4>;index=3D1.2.1;rc
> >>>>    History-Info: <sip:vm@example.com>;index=3D1.3;mp=3D1.2
> >>>>    History-Info: <sip:vm@192.0.2.5>;index=3D1.3.1
> >>>>
> >>>> If the voicemail server is configured to use the voicemail box
> >>>> of the initial target it can look for the first mp tag and look
> >>>> at the hi-entry that mp index references (bob@example.com).
> >>>>
> >>>> If the voicemail server is configured to use the voicemail box
> >>>> of the last target then it can look for the last mp tag and look
> >>>> at the hi-entry that mp index references (carol@example.com) etc.
> >>>>
> >>>> Regards
> >>>> Shida
> >>>>
> >>>> On Dec 1, 2009, at 7:56 AM, Francois AUDET wrote:
> >>>>
> >>>>> On Nov 30, 2009, at 5:13 , Alan Johnston wrote:
> >>>>>
> >>>>>> Does this document have a good voicemail example? =20
> This is the application I am most interested in, so it would=20
> be nice to have a very clear example of this.
> >>>>>>
> >>>>> That was the intent of section B.2 (in the appendix).
> >>>>>
> >>>>> Are you saying you'd want a better voicemail example?=20
> Or that we add more voicemail examples? What do you have in mind?
> >>>>> _______________________________________________
> >>>>> 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
> >>
> >=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =

From adam@estacado.net  Fri Dec  4 15:51:08 2009
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Date: Fri, 04 Dec 2009 17:49:43 -0600
From: Adam Roach <adam@estacado.net>
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Subject: [sipcore] Call for consensus: adopting draft-camarillo-sipcore-reinvite
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[as chair]

Regarding the following milestone on our charter:

   Invite Transaction Handling Correction to IESG (PS)

To fulfill this milestone, there are two potential problems that may 
need to be addressed. One of them relates to handling of 200-class 
responses to INVITE messages; the other relates to re-INVITE and target 
refresh handling.

The first problem is addressed by an adopted working group item 
(draft-ietf-sipcore-invfix). We do not currently have an adopted item 
for the second problem.

The chairs would like to solicit feedback from the group regarding 
adoption of draft-camarillo-sipcore-reinvite as a working group item to 
fix the second set of problems. Please comment on the SIPCORE mailing 
list. If you agree with this adoption, a simple note stating as much is 
encouraged: we cannot determine consensus without feedback, and silence 
is not very informative.

Thanks!

/a

From pkyzivat@cisco.com  Fri Dec  4 16:26:04 2009
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Subject: Re: [sipcore] Call for consensus: adopting	draft-camarillo-sipcore-reinvite
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yes, adopt draft-camarillo-sipcore-reinvite

	Thanks,
	Paul

Adam Roach wrote:
> [as chair]
> 
> Regarding the following milestone on our charter:
> 
>   Invite Transaction Handling Correction to IESG (PS)
> 
> To fulfill this milestone, there are two potential problems that may 
> need to be addressed. One of them relates to handling of 200-class 
> responses to INVITE messages; the other relates to re-INVITE and target 
> refresh handling.
> 
> The first problem is addressed by an adopted working group item 
> (draft-ietf-sipcore-invfix). We do not currently have an adopted item 
> for the second problem.
> 
> The chairs would like to solicit feedback from the group regarding 
> adoption of draft-camarillo-sipcore-reinvite as a working group item to 
> fix the second set of problems. Please comment on the SIPCORE mailing 
> list. If you agree with this adoption, a simple note stating as much is 
> encouraged: we cannot determine consensus without feedback, and silence 
> is not very informative.
> 
> Thanks!
> 
> /a
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 

From shida@ntt-at.com  Fri Dec  4 22:19:15 2009
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Cc: Adam Roach <adam@estacado.net>, SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Call for consensus: adopting	draft-camarillo-sipcore-reinvite
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+1
=20
 Regards,
  Shida

On Dec 5, 2009, at 9:25 AM, Paul Kyzivat wrote:

> yes, adopt draft-camarillo-sipcore-reinvite
>=20
> 	Thanks,
> 	Paul
>=20
> Adam Roach wrote:
>> [as chair]
>> Regarding the following milestone on our charter:
>>  Invite Transaction Handling Correction to IESG (PS)
>> To fulfill this milestone, there are two potential problems that may =
need to be addressed. One of them relates to handling of 200-class =
responses to INVITE messages; the other relates to re-INVITE and target =
refresh handling.
>> The first problem is addressed by an adopted working group item =
(draft-ietf-sipcore-invfix). We do not currently have an adopted item =
for the second problem.
>> The chairs would like to solicit feedback from the group regarding =
adoption of draft-camarillo-sipcore-reinvite as a working group item to =
fix the second set of problems. Please comment on the SIPCORE mailing =
list. If you agree with this adoption, a simple note stating as much is =
encouraged: we cannot determine consensus without feedback, and silence =
is not very informative.
>> Thanks!
>> /a
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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Thread-Topic: [sipcore] Call for consensus:	adopting draft-camarillo-sipcore-reinvite
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+1

> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf Of Paul Kyzivat
> Sent: Friday, December 04, 2009 7:26 PM
> To: Adam Roach
> Cc: SIPCORE
> Subject: Re: [sipcore] Call for consensus: adopting draft-camarillo-
> sipcore-reinvite
>=20
> yes, adopt draft-camarillo-sipcore-reinvite
>=20
> 	Thanks,
> 	Paul
>=20
> Adam Roach wrote:
> > [as chair]
> >
> > Regarding the following milestone on our charter:
> >
> >   Invite Transaction Handling Correction to IESG (PS)
> >
> > To fulfill this milestone, there are two potential problems that may
> > need to be addressed. One of them relates to handling of 200-class
> > responses to INVITE messages; the other relates to re-INVITE and
> target
> > refresh handling.
> >
> > The first problem is addressed by an adopted working group item
> > (draft-ietf-sipcore-invfix). We do not currently have an adopted item
> > for the second problem.
> >
> > The chairs would like to solicit feedback from the group regarding
> > adoption of draft-camarillo-sipcore-reinvite as a working group item
> to
> > fix the second set of problems. Please comment on the SIPCORE mailing
> > list. If you agree with this adoption, a simple note stating as much
> is
> > encouraged: we cannot determine consensus without feedback, and
> silence
> > is not very informative.
> >
> > Thanks!
> >
> > /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

From gonzalo.camarillo@ericsson.com  Sat Dec  5 22:37:58 2009
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Cc: SIPCORE <sipcore@ietf.org>, wang.libo@zte.com.cn
Subject: Re: [sipcore] Suspending and Resuming modification of session or stream? //New revision of the re-INVITE handling draft
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Hi,

[I am resurrecting this old thread because it did not reach a conclusion.]

Originally, preconditions were designed with initial INVITE requests in 
mind. The idea was to keep the UAS from alerting the callee until the 
media streams with preconditions were ready to be established (e.g., 
their QoS was in place). That is, if there were some streams that did 
not have preconditions, they would not be established until the 
preconditions for the other streams were met. For example, let us assume 
an initial INVITE with an audio stream without preconditions and a video 
stream with QoS preconditions. Until the QoS for the video stream is in 
place, the audio stream will not be established and the callee will not 
be alerted. In short, when one or more streams within a session have 
preconditions, the whole session establishment is suspended until those 
preconditions are met.

When we extended the concept of preconditions to re-INVITEs, we wanted 
to have similar behavior for initial INVITEs and for re-INVITEs. That is 
why RFC 4032 talks about about *session* establishment being suspended:

http://tools.ietf.org/html/rfc4032#section-3.4

Therefore, the idea was that no changes were applied to any stream until 
all the preconditions were met.

So, the above was our original thinking. However, at present we have a 
better understanding about the problems with long-lived re-INVITE 
transactions. If we allow the whole session establishment to be 
suspended, it would be impossible to change the session parameters of 
any stream while the UAs were working on meeting the preconditions of 
other streams. On the other hand, we want to be able to make the changes 
to one stream conditional to the changes on other stream being executed 
(e.g., use this new audio codec if you accept adding a video stream to 
the session; otherwise, keep on using the old audio codec).

Before we explore this more deeply, I would like to understand if there 
is interest in resolving this type of issue. Some time ago I put 
together a draft dealing with a similar issue (media state during 
preconditions) but we concluded that, at that point, there was not 
enough interest for me (or anyone else for that matter) to put more 
energy into it:

http://tools.ietf.org/html/draft-camarillo-sipping-precons-00

If people want these issues to be clarified, we should by all means do 
so. However, if people do not really care about these corner cases, we 
should prioritize other specs.

Thanks,

Gonzalo



gao.yang2@zte.com.cn wrote:
> 
> Considering a use case, A and B has a audio only session. Then A send a 
> Re-INVITE for adding vedio stream( with precondition) and changing codec 
> or address for audio stream( without preconditon).
> 
> If suspending is effective for whole session, then streams without 
> precondition should use old parameters while in suspending state.
> For the use case, A and B should not use new codec or address for audio 
> before vedio's preconditon is met.
> 
> If suspending is effective for streams with precondition, the then 
> streams without preconditio can use new parameters after first O/A 
> during the Re-INVITE.
> For the use case, A and B can use new codec or address for audio before 
> vedio's preconditon is met.
> 
> 
> I prefer the first one, as;
> 1. To delay using of new parameters for stream without precondition may 
> arose media clip problem.
> 2. Behavior of streams without precondition should be independent of 
> existence of other streams( with precondition).
> 
> ===================================
> Zip    : 210012
> Tel    : 87211
> Tel2   :(+86)-025-52877211
> e_mail : gao.yang2@zte.com.cn
> ===================================
> 
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From gonzalo.camarillo@ericsson.com  Sun Dec  6 08:40:07 2009
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Subject: Re: [sipcore] Some comments on draft-barnes-sipcore-rfc4244bis-03
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Hi,

> When you think about it, it makes sense.
> 
> If you call me, you would expect to get my voicemail, not somebody else's.

yes, Paul's example has been discussed *many* times. So, addressing it 
in some way would be a good thing.

Cheers,

Gonzalo


From gonzalo.camarillo@ericsson.com  Sun Dec  6 09:11:10 2009
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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: Re: [sipcore] Reviewing the sec-flows draft
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Folks,

as you know, when we restructured the RAI area one of the plans was to 
avoid working on specifications people were not interested in any 
longer. Adam and I have had to assign a new editor (with cycles) to this 
document and have unsuccessfully asked for reviewers *many* times. If we 
do not see some activity related to this draft in the near future the 
only conclusion we can draw is that we should abandon it.

So, if you think this is important, now it would be a good time for 
volunteering. Also, if you think we should abandon this draft, let us 
know as well.

Thanks,

Gonzalo
SIPCORE co-chair


Gonzalo Camarillo wrote:
> Folks,
> 
> as you may have noticed, Brian has submitted a new revision of this 
> draft and sent a couple of notes to the list (on November 24th) 
> describing the topics on which he expects to get feedback. Please, send 
> your comments to this list.
> 
> Thanks,
> 
> Gonzalo
> SIPCORE co-chair
> 
> Gonzalo Camarillo wrote:
>> Folks,
>>
>> as you know, we have asked for volunteers to review the sec-flows 
>> draft many times. We got a couple of reviews but we would like to get 
>> at least one dedicated review more. If you want to volunteer, please 
>> send your review to the list.
>>
>> Additionally, the reviews we got pointed out at issues that require WG 
>> consensus (as opposed to the editor simply going off and implementing 
>> them).
>>
>> As next steps, the draft's editor (Brian) will be submitting a new 
>> revision of the draft because the current one has expired. Then, he 
>> will be sending a set of emails to the list so that we can reach 
>> consensus on the issues that need discussion.
>>
>> Thanks,
>>
>> Gonzalo
>> SIPCORE co-chair
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>
> 
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 


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Cc: Francois Audet <audet.francois@gmail.com>, sipcore@ietf.org, R.Jesske@telekom.de
Subject: Re: [sipcore] Some comments on draft-barnes-sipcore-rfc4244bis-03
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The main tactic I know of that is currently deployed to address this 
need is to cancel the forwarded call after a few "rings", hopefully 
before it is transferred to voicemail there. But of course that doesn't 
work if you guess wrong about the number of rings, and especially if the 
call goes direct to voicemail.

While there is no way, once I have forwarded a call to you, for me to 
*prevent* you from handling the call as voicemail, it is typically in 
the best interests of both of us. Specifically, if I am forwarding the 
call to you, but asked you *not* to send the call to your voicemail, 
then honoring that saves you the trouble handling the voicemail, and 
also keeps you in my good graces.

So the key here is to communicate preferences for how the call should be 
handed, and define the recommended handling of calls in the presence of 
such a recommendation.

The basics to do this have been in callerprefs for a long time. But I 
don't think they have been considered seriously in this context. I'm not 
*certain* that is sufficient to address the problem, but it seems like 
the right starting point.

Note that this sort of approach *does not* require looking back in the 
History-Info to decide which voice mailbox to use.

	Thanks,
	Paul


Gonzalo Camarillo wrote:
> Hi,
> 
>> When you think about it, it makes sense.
>>
>> If you call me, you would expect to get my voicemail, not somebody 
>> else's.
> 
> yes, Paul's example has been discussed *many* times. So, addressing it 
> in some way would be a good thing.
> 
> Cheers,
> 
> Gonzalo
> 
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 

From christer.holmberg@ericsson.com  Sun Dec  6 10:40:02 2009
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 + 1=20

-----Original Message-----
From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
Behalf Of Paul Kyzivat
Sent: 5. joulukuuta 2009 2:26
To: Adam Roach
Cc: SIPCORE
Subject: Re: [sipcore] Call for consensus:adopting
draft-camarillo-sipcore-reinvite

yes, adopt draft-camarillo-sipcore-reinvite

	Thanks,
	Paul

Adam Roach wrote:
> [as chair]
>=20
> Regarding the following milestone on our charter:
>=20
>   Invite Transaction Handling Correction to IESG (PS)
>=20
> To fulfill this milestone, there are two potential problems that may=20
> need to be addressed. One of them relates to handling of 200-class=20
> responses to INVITE messages; the other relates to re-INVITE and=20
> target refresh handling.
>=20
> The first problem is addressed by an adopted working group item=20
> (draft-ietf-sipcore-invfix). We do not currently have an adopted item=20
> for the second problem.
>=20
> The chairs would like to solicit feedback from the group regarding=20
> adoption of draft-camarillo-sipcore-reinvite as a working group item=20
> to fix the second set of problems. Please comment on the SIPCORE=20
> mailing list. If you agree with this adoption, a simple note stating=20
> as much is
> encouraged: we cannot determine consensus without feedback, and=20
> silence is not very informative.
>=20
> Thanks!
>=20
> /a
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>=20
_______________________________________________
sipcore mailing list
sipcore@ietf.org
https://www.ietf.org/mailman/listinfo/sipcore

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From vkg@bell-labs.com  Sun Dec  6 18:41:22 2009
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On 12/06/2009 11:10 AM, Gonzalo Camarillo wrote:
> Folks,
>
> as you know, when we restructured the RAI area one of the plans was to
> avoid working on specifications people were not interested in any
> longer. Adam and I have had to assign a new editor (with cycles) to this
> document and have unsuccessfully asked for reviewers *many* times. If we
> do not see some activity related to this draft in the near future the
> only conclusion we can draw is that we should abandon it.
>
> So, if you think this is important, now it would be a good time for
> volunteering. Also, if you think we should abandon this draft, let us
> know as well.

We should NOT abandon it; the draft has some good information.
I had promised Brian a review of some sections after Thanksgiving,
but have been tied up last week.  I intend to send out comments
this week (i.e., Dec. 7).

Thanks,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
Web:   http://ect.bell-labs.com/who/vkg/

From salvatore.loreto@ericsson.com  Mon Dec  7 00:43:12 2009
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Cc: Adam Roach <adam@estacado.net>, SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Call for	consensus:adopting	draft-camarillo-sipcore-reinvite
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+1
I think we need the WG item

/Sal
> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf Of Paul Kyzivat
> Sent: 5. joulukuuta 2009 2:26
> To: Adam Roach
> Cc: SIPCORE
> Subject: Re: [sipcore] Call for consensus:adopting
> draft-camarillo-sipcore-reinvite
>
> yes, adopt draft-camarillo-sipcore-reinvite
>
> 	Thanks,
> 	Paul
>
> Adam Roach wrote:
>   
>> [as chair]
>>
>> Regarding the following milestone on our charter:
>>
>>   Invite Transaction Handling Correction to IESG (PS)
>>
>> To fulfill this milestone, there are two potential problems that may 
>> need to be addressed. One of them relates to handling of 200-class 
>> responses to INVITE messages; the other relates to re-INVITE and 
>> target refresh handling.
>>
>> The first problem is addressed by an adopted working group item 
>> (draft-ietf-sipcore-invfix). We do not currently have an adopted item 
>> for the second problem.
>>
>> The chairs would like to solicit feedback from the group regarding 
>> adoption of draft-camarillo-sipcore-reinvite as a working group item 
>> to fix the second set of problems. Please comment on the SIPCORE 
>> mailing list. If you agree with this adoption, a simple note stating 
>> as much is
>> encouraged: we cannot determine consensus without feedback, and 
>> silence is not very informative.
>>
>> Thanks!
>>
>> /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
>   


From rjsparks@nostrum.com  Mon Dec  7 11:32:50 2009
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Subject: Re: [sipcore] AD review draft-ietf-sipcore-199-00
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On Oct 14, 2009, at 1:20 PM, Christer Holmberg wrote:

>
> Hi Robert,
>
> Comments inline.
>
> -------------------
>
>> In preparing this document for IETF last call, I found a few points  
>> we
> should discuss before moving forward.
>>
>> Major Comments
>>
>> I don't understand the sentence in section 4.1 that talks about
> RFC5009.  There is not enough here to inform an
>> implementor and don't think 5009 is required reading to understand or
> implement this document (hence, I don't agree that
>> 5009 should be a normative reference). The sentence has been there  
>> for
> a long time, but I think other things have pulled
>> reviewer's eyes away from it.
>
> I can change 5009 into an informative reference.
>

You didn't do this in -01?

RjS


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Subject: Re: [sipcore] AD review draft-ietf-sipcore-199-00
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On Oct 14, 2009, at 1:20 PM, Christer Holmberg wrote:

...

>
>
>> In Sections 5 and 9, the document says to include SDP in a 199 if
>> RFC3263 requires it. I would like to see some additional discussion
> around whether this is the right way to deal with
>> this edge case. If it is, then this document needs to explain how a  
>> UA
> should handle receiving an offer in a 199.  A
>> different path out is to say that such a 199 MUST NOT be sent  
>> reliably.
> (More explanation of why we allow 199 to be sent
>> reliably at all (why this is SHOULD not MUST) would also help).
>
> I wouldn't like to generally disallow to send 199 reliably, in case
> there is some future use-case which would need it. That's the reason  
> for
> the SHOULD: don't send it reliably unless you have a good reason to do
> so.
>
> Regarding the specific must-include-SDP-offer case (which will be very
> rare, in my opinion) I see two alternatives:
>
> 1.  We disallow reliable 199 in such cases
> 2.  We say that the UAS must include e.g. an SDP offer without any m-
> lines.
>
> My proposal is to describe both alternatives, and say that the UAS  
> MUST
> use of them.
>

I'm not seeing any changes in -01 addressing this comment?


RjS

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On Oct 14, 2009, at 1:20 PM, Christer Holmberg wrote:
...

>
>> Last sentence on page 8: This paragraph is talking about the proxy
> generating a 199 itself. It would be terribly
>> misleading from a diagnostics point of view for the proxy to put the
> Record-Route and contact from some UA (it would then
>> look like the 199 came from the remote endpoint - is that what you  
>> are
> trying to accomplish?). Why is having these bits
>> in the message useful at all?
>
> The intention is NOT to make the 199 look like it came from the
> endpoint, but someone commented that the UA could be confused if the
> bits aren't included.
>
> Personally I would bo ok to remove the text, since the proxy is not
> allowed to send a 199 reliably (there will be no PRACK etc, for which
> the bits would b needed)

The document shows no change from -00 addressing this comment either?

RjS


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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><br><div><div>On Oct 14, 2009, =
at 1:20 PM, Christer Holmberg =
wrote:</div><div>...</div><div><br></div><blockquote =
type=3D"cite"><div><font class=3D"Apple-style-span" =
color=3D"#000000"><br></font><blockquote type=3D"cite">Last sentence on =
page 8: This paragraph is talking about the =
proxy<br></blockquote>generating a 199 itself. It would be terribly =
<br><blockquote type=3D"cite">misleading from a diagnostics point of =
view for the proxy to put the<br></blockquote>Record-Route and contact =
from some UA (it would then <br><blockquote type=3D"cite">look like the =
199 came from the remote endpoint - is that what you =
are<br></blockquote>trying to accomplish?). Why is having these bits =
<br><blockquote type=3D"cite">in the message useful at =
all?<br></blockquote><br>The intention is NOT to make the 199 look like =
it came from the<br>endpoint, but someone commented that the UA could be =
confused if the<br>bits aren't included.<br><br>Personally I would bo ok =
to remove the text, since the proxy is not<br>allowed to send a 199 =
reliably (there will be no PRACK etc, for which<br>the bits would b =
needed)<br></div></blockquote><div><br></div><div>The document shows no =
change from -00 addressing this comment =
either?</div><div><br></div><div>RjS</div><div><br></div></div></body></ht=
ml>=

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Subject: Re: [sipcore] AD review draft-ietf-sipcore-199-00 - 199 after final response
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I still can't find where in the document this is answered. Can you  
point to the sentence(s) that you
think covers these.

RjS

On Oct 16, 2009, at 1:29 PM, Christer Holmberg wrote:

>
> Hi,
>
>>> I don't think the document currently provides a solid answer to this
>>> question (maybe I'm missing it): If a proxy has
>>> multiple branches outstanding and receives a 200 OK on one of  
>>> them, it
>>> forwards it immediately. As the other branches
>>> complete, should the proxy send 199s for them (assuming they didn't
>>> complete with additional 200s)?
>>
>> The intention is that NO 199s are sent after 200 OK has been  
>> forwarded
> on a branch, and the proxy sends CANCEL on the other branches.
>>
>> I DO agree that it is not clear in the spec, and that we can add some
> text about that.
>
> Actually, when taking a closer look at the text, I think this is  
> already
> covered. The text says that when the proxy receives a non-2xx final
> response, it sends 199 if the forking proxy has not already sent a  
> final
> resposne for any of the early dialogs. So, if a final response (2xx or
> non-2xx) response has been sent, additional non-2xx responses will not
> trigger 199.
>
> Regards,
>
> Christer
>


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While these were less important, I'm not seeing any changes at all  
addressing them, even for those you indicated you would make changes.

At this point, I'd like to double-check that you submitted what you  
meant to submit as -01?

RjS

On Oct 14, 2009, at 1:20 PM, Christer Holmberg wrote:
>
>> Minor Comments
>>
>> Bottom of page 3, next to last paragraph, last sentence: This assumes
> the response can't be recovered from, precluding
>> responding to authentication challenges, extension negotiation, etc.
>
> Again, I am not sure I understand the comment.
>
> -------------------
>
>> Page 6 - NOTE in section 4.1: Why isn't this calling out DTLS/SRTP?
>
> I can add it. How would you like to change to look like?
>
> -------------------
>
>> Section 8 is completely redundant. It adds no value to the draft. I
> suggest deleting it.
>
> I based this on other drafts, but I can remove it.
>
> -------------------
>
>> Nits
>>
>> The 2119 language (two instances of MAY unless I missed more)  
>> appearing
> in the introduction are inappropriate in an
>> introduction.
>
> I propose to change "MAY" to "can" (in the introduction).
>
> -------------------
>
>> Page 5, second paragraph : is "shall not" meant to be normative?
>
> Yes. I'll change it to uppercase.
>
> -------------------
>
>> Page 8 Section 6 first paragraph: Suggest dropping "on an early  
>> dialog"
> from the first sentence. Does it add anything?
>> This would help avoid the problem the current text has about assuming
> proxies know about dialogs.
>
> I can drop it from the first sentence.
>
> -------------------
>
> Thanks for your comments!
>
> Regards,
>
> Christer


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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><div>While these were less =
important, I'm not seeing any changes at all addressing them, even for =
those you indicated you would make changes.</div><div><br></div><div>At =
this point, I'd like to double-check that you submitted what you meant =
to submit as =
-01?</div><div><br></div><div>RjS</div><div><br></div><div><div>On Oct =
14, 2009, at 1:20 PM, Christer Holmberg wrote:</div><blockquote =
type=3D"cite"><div><font class=3D"Apple-style-span" =
color=3D"#000000"><br></font><blockquote type=3D"cite">Minor =
Comments<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">Bottom of page =
3, next to last paragraph, last sentence: This =
assumes<br></blockquote>the response can't be recovered from, precluding =
<br><blockquote type=3D"cite">responding to authentication challenges, =
extension negotiation, etc.<br></blockquote><br>Again, I am not sure I =
understand the comment.<br><br>-------------------<br><br><blockquote =
type=3D"cite">Page 6 - NOTE in section 4.1: Why isn't this calling out =
DTLS/SRTP?<br></blockquote><br>I can add it. How would you like to =
change to look like?<br><br>-------------------<br><br><blockquote =
type=3D"cite">Section 8 is completely redundant. It adds no value to the =
draft. I<br></blockquote>suggest deleting it.<br><br>I based this on =
other drafts, but I can remove =
it.<br><br>-------------------<br><br><blockquote =
type=3D"cite">Nits<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">The 2119 =
language (two instances of MAY unless I missed more) =
appearing<br></blockquote>in the introduction are inappropriate in an =
<br><blockquote type=3D"cite">introduction.<br></blockquote><br>I =
propose to change "MAY" to "can" (in the =
introduction).<br><br>-------------------<br><br><blockquote =
type=3D"cite">Page 5, second paragraph : is "shall not" meant to be =
normative?<br></blockquote><br>Yes. I'll change it to =
uppercase.<br><br>-------------------<br><br><blockquote =
type=3D"cite">Page 8 Section 6 first paragraph: Suggest dropping "on an =
early dialog"<br></blockquote>from the first sentence. Does it add =
anything? <br><blockquote type=3D"cite">This would help avoid the =
problem the current text has about assuming<br></blockquote>proxies know =
about dialogs.<br><br>I can drop it from the first =
sentence.<br><br>-------------------<br><br>Thanks for your =
comments!<br><br>Regards,<br><br>Christer<br></div></blockquote></div><br>=
</body></html>=

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=20
Hi Robert,

It seems like I did some of the changes in another XML file than the one
I used to produce the -01 draft - sorry for that :(

>>>I don't understand the sentence in section 4.1 that talks about
>>>RFC5009. There is not enough here to inform an
>>>implementor and don't think 5009 is required reading to understand or
>>>implement this document (hence, I don't agree that
>>>5009 should be a normative reference). The sentence has been there=20
>>>for a long time, but I think other things have pulled
>>>reviewer's eyes away from it.
>>
>>I can change 5009 into an informative reference.
>
>You didn't do this in -01?

It is now in the correct XML file, so it will be done in -02.

Regards,

Christer



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Hi,
=20
>While these were less important, I'm not seeing any changes at all
addressing them, even for those you indicated you would make changes.
>
>
>At this point, I'd like to double-check that you submitted what you
meant to submit as -01?

Again, most changes were done (note the DTLS/SRTP issue, though), but in
the wrong xml file. "Change done now" means that the changes will be in
version -02.

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


>>Bottom of page 3, next to last paragraph, last sentence: This assumes
>>the response can't be recovered from, precluding responding to
authentication challenges, extension negotiation, etc.
>	=09
>Again, I am not sure I understand the comment.

The sentence tries to describe current behavior. I added ", or retry,"
to the text, in order to clarify that the UAC may retry the session
setup.

"When SIP entities upstream receive the non-2xx final response they will
release resources associated with the
session, and the UAC will terminate, or retry, the session setup."
=09

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

=09
>>Page 6 - NOTE in section 4.1: Why isn't this calling out DTLS/SRTP?
>	=09
>I can add it. How would you like to change to look like?

On this topic I did send out an e-mail (RE: [sipcore] AD review
draft-ietf-sipcore-199-00 - DTLS/SRTP), where I said:

"I would need some clarification on why DTLS would have to be added
here. The issue is not related to a specific key management mechanism
used, but with the fact that the media is encrypted in general.

Of course different key management mechanisms may have different impacts
on the needed resources etc, but I still don't see why this would be an
issue specific to DTLS."

=09
-------------------

=09
>>Section 8 is completely redundant. It adds no value to the draft. I
suggest deleting it.
>
>I based this on other drafts, but I can remove it.

Change done now.

=09
-------------------

=09
>>The 2119 language (two instances of MAY unless I missed more)
appearing
>>in the introduction are inappropriate in an introduction.
>	=09
>I propose to change "MAY" to "can" (in the introduction).

Change done now.
=09

-------------------
=09
=09
>>Page 5, second paragraph : is "shall not" meant to be normative?
>	=09
>Yes. I'll change it to uppercase.
=09
Change done now.

-------------------
=09
=09
>>Page 8 Section 6 first paragraph: Suggest dropping "on an early
dialog"
>>from the first sentence. Does it add anything?=20
>>This would help avoid the problem the current text has about assuming
>>proxies know about dialogs.
>=09
>I can drop it from the first sentence.

Change done now.
=09
-------------------


Again, sorry for the mess.
=09
=09
Regards,
=09
Christer
=09



From christer.holmberg@ericsson.com  Tue Dec  8 12:50:52 2009
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Subject: Re: [sipcore] AD review draft-ietf-sipcore-199-00 - 199 after final response
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Hi,=20

>I still can't find where in the document this is answered. Can you
point to the sentence(s) that you think covers these.

Section 6 says:

"When a forking proxy receives a non-2xx final response that it
recognizes as terminating one or more early dialogs, it SHOULD
generate and send a 199 response upstream for each of the terminated
early dialogs that satisfy each of the following conditions:

- the forking proxy does not intend to forward the final response
immediately (in accordance with rules for a forking proxy)

- the UAC has indicated support (using the 199 option tag) for the
199 response code

- the forking proxy has not already received and forwarded a 199
response for that early dialog

- the forking proxy has not already sent a final response for any of
the early dialogs"

I think the fourth condition addresses your issue.

Regards,

Christer




On Oct 16, 2009, at 1:29 PM, Christer Holmberg wrote:

>
> Hi,
>
>>> I don't think the document currently provides a solid answer to this

>>> question (maybe I'm missing it): If a proxy has multiple branches=20
>>> outstanding and receives a 200 OK on one of them, it forwards it=20
>>> immediately. As the other branches complete, should the proxy send=20
>>> 199s for them (assuming they didn't complete with additional 200s)?
>>
>> The intention is that NO 199s are sent after 200 OK has been=20
>> forwarded
> on a branch, and the proxy sends CANCEL on the other branches.
>>
>> I DO agree that it is not clear in the spec, and that we can add some
> text about that.
>
> Actually, when taking a closer look at the text, I think this is=20
> already covered. The text says that when the proxy receives a non-2xx=20
> final response, it sends 199 if the forking proxy has not already sent

> a final resposne for any of the early dialogs. So, if a final response

> (2xx or
> non-2xx) response has been sent, additional non-2xx responses will not

> trigger 199.
>
> Regards,
>
> Christer
>


From christer.holmberg@ericsson.com  Tue Dec  8 12:55:19 2009
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Hi,

>>Last sentence on page 8: This paragraph is talking about the proxy
>>generating a 199 itself. It would be terribly=20
>>misleading from a diagnostics point of view for the proxy to put the
>>Record-Route and contact from some UA (it would then look like the 199
came=20
>>from the remote endpoint - is that what you are trying to
accomplish?).=20
>>Why is having these bits in the message useful at all?
>	=09
>The intention is NOT to make the 199 look like it came from the
>endpoint, but someone commented that the UA could be confused if the
>bits aren't included.
>=09
>Personally I would bo ok to remove the text, since the proxy is not
>allowed to send a 199 reliably (there will be no PRACK etc, for which
>the bits would b needed)
>=09
>The document shows no change from -00 addressing this comment either?

In version -02 the last sentence, talking about Contact and
Record-Route, will be removed.

Regards,

Christer

From rjsparks@nostrum.com  Tue Dec  8 13:03:44 2009
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Subject: Re: [sipcore] AD review draft-ietf-sipcore-199-00 - 199 after final response
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Ok - I can see how you are reading the text.

I think the opportunity for mis-interpretation is very high here  
though, and
encourage adding a sentence after this saying something like:

"As a consequence, once a final response to the INVITE has been issued  
by
  the proxy, no further 199 messages corresponding to the INVITE will be
  generated or forwarded by the proxy"

RjS

On Dec 8, 2009, at 2:49 PM, Christer Holmberg wrote:

>
> Hi,
>
>> I still can't find where in the document this is answered. Can you
> point to the sentence(s) that you think covers these.
>
> Section 6 says:
>
> "When a forking proxy receives a non-2xx final response that it
> recognizes as terminating one or more early dialogs, it SHOULD
> generate and send a 199 response upstream for each of the terminated
> early dialogs that satisfy each of the following conditions:
>
> - the forking proxy does not intend to forward the final response
> immediately (in accordance with rules for a forking proxy)
>
> - the UAC has indicated support (using the 199 option tag) for the
> 199 response code
>
> - the forking proxy has not already received and forwarded a 199
> response for that early dialog
>
> - the forking proxy has not already sent a final response for any of
> the early dialogs"
>
> I think the fourth condition addresses your issue.
>
> Regards,
>
> Christer
>
>
>
>
> On Oct 16, 2009, at 1:29 PM, Christer Holmberg wrote:
>
>>
>> Hi,
>>
>>>> I don't think the document currently provides a solid answer to  
>>>> this
>
>>>> question (maybe I'm missing it): If a proxy has multiple branches
>>>> outstanding and receives a 200 OK on one of them, it forwards it
>>>> immediately. As the other branches complete, should the proxy send
>>>> 199s for them (assuming they didn't complete with additional 200s)?
>>>
>>> The intention is that NO 199s are sent after 200 OK has been
>>> forwarded
>> on a branch, and the proxy sends CANCEL on the other branches.
>>>
>>> I DO agree that it is not clear in the spec, and that we can add  
>>> some
>> text about that.
>>
>> Actually, when taking a closer look at the text, I think this is
>> already covered. The text says that when the proxy receives a non-2xx
>> final response, it sends 199 if the forking proxy has not already  
>> sent
>
>> a final resposne for any of the early dialogs. So, if a final  
>> response
>
>> (2xx or
>> non-2xx) response has been sent, additional non-2xx responses will  
>> not
>
>> trigger 199.
>>
>> Regards,
>>
>> Christer
>>
>


From christer.holmberg@ericsson.com  Tue Dec  8 13:06:22 2009
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=20
Hi,

>>In Sections 5 and 9, the document says to include SDP in a 199 if
>>RFC3263 requires it. I would like to see some additional discussion
>>around whether this is the right way to deal with
>>this edge case. If it is, then this document needs to explain how a=20
>>UA should handle receiving an offer in a 199.  A
>>different path out is to say that such a 199 MUST NOT be sent=20
>>reliably.
>>(More explanation of why we allow 199 to be sent
>>reliably at all (why this is SHOULD not MUST) would also help).
>>
>>I wouldn't like to generally disallow to send 199 reliably, in case=20
>>there is some future use-case which would need it. That's the reason=20
>>for the SHOULD: don't send it reliably unless you have a good reason=20
>>to do so.
>>
>>Regarding the specific must-include-SDP-offer case (which will be very

>>rare, in my opinion) I see two alternatives:
>>
>>1.  We disallow reliable 199 in such cases 2.  We say that the UAS=20
>>must include e.g. an SDP offer without any m- lines.
>>
>>My proposal is to describe both alternatives, and say that the UAS=20
>>MUST use of them.
>
>
>I'm not seeing any changes in -01 addressing this comment?

I changed and extended (will be seen in -02) the note into normative
text:

"A 199 Early Dialog Terminated provisional response SHOULD NOT contain
an SDP offer/answer message body, unless required by the rules in
[RFC3264].

If the INVITE request did not contain an SDP offer, and the 199
response is the first reliably sent response, the 199 response is
required to contain an SDP offer. In this case the UAS SHOULD send
the 199 response unreliable, or include an SDP offer with no m- lines=20
in the reliable 199 response."

Regards,

Christer


From christer.holmberg@ericsson.com  Tue Dec  8 13:15:23 2009
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Thread-Topic: [sipcore] AD review draft-ietf-sipcore-199-00 - 199 after final response
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References: <70463F72-4486-4384-B294-078EA191CC9B@nostrum.com> <CA9998CD4A020D418654FCDEF4E707DF0B16853D@esealmw113.eemea.ericsson.se> <CA9998CD4A020D418654FCDEF4E707DF0B168560@esealmw113.eemea.ericsson.se> <3E7D74BA-84DB-4906-BB66-C30080071F5A@nostrum.com> <CA9998CD4A020D418654FCDEF4E707DF0FAF2CD1@esealmw113.eemea.ericsson.se> <06662B84-BB4C-49F1-A3D4-738A60712247@nostrum.com>
From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Robert Sparks" <rjsparks@nostrum.com>
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Subject: Re: [sipcore] AD review draft-ietf-sipcore-199-00 - 199 after final response
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Hi,=20

>Ok - I can see how you are reading the text.
>
>I think the opportunity for mis-interpretation is very high here
though, and encourage adding a sentence after this saying something
like:
>
>"As a consequence, once a final response to the INVITE has been issued
by
>the proxy, no further 199 messages corresponding to the INVITE will be
>generated or forwarded by the proxy"

I can do that. I will add the text below the conditions.

Regards,

Christer



On Dec 8, 2009, at 2:49 PM, Christer Holmberg wrote:

>
> Hi,
>
>> I still can't find where in the document this is answered. Can you
> point to the sentence(s) that you think covers these.
>
> Section 6 says:
>
> "When a forking proxy receives a non-2xx final response that it=20
> recognizes as terminating one or more early dialogs, it SHOULD=20
> generate and send a 199 response upstream for each of the terminated=20
> early dialogs that satisfy each of the following conditions:
>
> - the forking proxy does not intend to forward the final response=20
> immediately (in accordance with rules for a forking proxy)
>
> - the UAC has indicated support (using the 199 option tag) for the
> 199 response code
>
> - the forking proxy has not already received and forwarded a 199=20
> response for that early dialog
>
> - the forking proxy has not already sent a final response for any of=20
> the early dialogs"
>
> I think the fourth condition addresses your issue.
>
> Regards,
>
> Christer
>
>
>
>
> On Oct 16, 2009, at 1:29 PM, Christer Holmberg wrote:
>
>>
>> Hi,
>>
>>>> I don't think the document currently provides a solid answer to=20
>>>> this
>
>>>> question (maybe I'm missing it): If a proxy has multiple branches=20
>>>> outstanding and receives a 200 OK on one of them, it forwards it=20
>>>> immediately. As the other branches complete, should the proxy send=20
>>>> 199s for them (assuming they didn't complete with additional 200s)?
>>>
>>> The intention is that NO 199s are sent after 200 OK has been=20
>>> forwarded
>> on a branch, and the proxy sends CANCEL on the other branches.
>>>
>>> I DO agree that it is not clear in the spec, and that we can add=20
>>> some
>> text about that.
>>
>> Actually, when taking a closer look at the text, I think this is=20
>> already covered. The text says that when the proxy receives a non-2xx

>> final response, it sends 199 if the forking proxy has not already=20
>> sent
>
>> a final resposne for any of the early dialogs. So, if a final=20
>> response
>
>> (2xx or
>> non-2xx) response has been sent, additional non-2xx responses will=20
>> not
>
>> trigger 199.
>>
>> Regards,
>>
>> Christer
>>
>


From gonzalo.camarillo@ericsson.com  Wed Dec  9 07:03:55 2009
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Subject: Re: [sipcore] AD review draft-ietf-sipcore-199-00
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Hi Christer,

> Again, most changes were done (note the DTLS/SRTP issue, though), but in
> the wrong xml file. "Change done now" means that the changes will be in
> version -02.

why don't you submit 02 now so that people can follow the discussions 
more clearly? You can submit 03 later if more changes are needed.

Thanks,

Gonzalo


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Forwarding in case you're not watching the SIPPING list closely:


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> From: The IESG <iesg-secretary@ietf.org>
> Date: December 8, 2009 3:44:44 PM CST
> To: IETF-Announce <ietf-announce@ietf.org>
> Cc: sipping@ietf.org
> Subject: [Sipping] Last Call: draft-ietf-sipping-config-framework (A  
> Framework for Session Initiation Protocol User Agent Profile  
> Delivery) to Proposed Standard
> Reply-To: ietf@ietf.org
>
> The IESG has received a request from the Session Initiation Proposal
> Investigation WG (sipping) to consider the following document:
>
> - 'A Framework for Session Initiation Protocol User Agent Profile
>   Delivery '
>   <draft-ietf-sipping-config-framework-16.txt> as a Proposed Standard
>
> The IESG plans to make a decision in the next few weeks, and solicits
> final comments on this action.  Please send substantive comments to  
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> The IPR disclosure at
> https://datatracker.ietf.org/ipr/search/?option=document_search&id_document_tag=10128
> was received after the working group requested publication. Please  
> consider that
> disclosure when preparing comments.
>
> The file can be obtained via
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>
>
> IESG discussion can be tracked via
> https://datatracker.ietf.org/public/pidtracker.cgi?command=view_id&dTag=10128&rfc_flag=0
>
> _______________________________________________
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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
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#000000"><b>From: </b></font><font face=3D"Helvetica" size=3D"3" =
style=3D"font: 12.0px Helvetica">The IESG &lt;<a =
href=3D"mailto:iesg-secretary@ietf.org">iesg-secretary@ietf.org</a>&gt;</f=
ont></div><div style=3D"margin-top: 0px; margin-right: 0px; =
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size=3D"3" color=3D"#000000" style=3D"font: 12.0px Helvetica; color: =
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style=3D"font: 12.0px Helvetica">December 8, 2009 3:44:44 PM =
CST</font></div><div style=3D"margin-top: 0px; margin-right: 0px; =
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style=3D"font: 12.0px Helvetica">IETF-Announce &lt;<a =
href=3D"mailto:ietf-announce@ietf.org">ietf-announce@ietf.org</a>&gt;</fon=
t></div><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: =
0px; margin-left: 0px; "><font face=3D"Helvetica" size=3D"3" =
color=3D"#000000" style=3D"font: 12.0px Helvetica; color: =
#000000"><b>Cc: </b></font><font face=3D"Helvetica" size=3D"3" =
style=3D"font: 12.0px Helvetica"><a =
href=3D"mailto:sipping@ietf.org">sipping@ietf.org</a></font></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><font face=3D"Helvetica" size=3D"3" color=3D"#000000" =
style=3D"font: 12.0px Helvetica; color: #000000"><b>Subject: =
</b></font><font face=3D"Helvetica" size=3D"3" style=3D"font: 12.0px =
Helvetica"><b>[Sipping] Last Call: draft-ietf-sipping-config-framework =
(A Framework for Session Initiation Protocol User Agent Profile =
Delivery) to Proposed Standard</b></font></div><div style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; "><font =
face=3D"Helvetica" size=3D"3" color=3D"#000000" style=3D"font: 12.0px =
Helvetica; color: #000000"><b>Reply-To: </b></font><font =
face=3D"Helvetica" size=3D"3" style=3D"font: 12.0px Helvetica"><a =
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style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; min-height: 14px; "><br></div> </div><div>The IESG has =
received a request from the Session Initiation Proposal =
<br>Investigation WG (sipping) to consider the following =
document:<br><br>- 'A Framework for Session Initiation Protocol User =
Agent Profile <br> &nbsp;&nbsp;Delivery '<br> =
&nbsp;&nbsp;&lt;draft-ietf-sipping-config-framework-16.txt&gt; as a =
Proposed Standard<br><br>The IESG plans to make a decision in the next =
few weeks, and solicits<br>final comments on this action. &nbsp;Please =
send substantive comments to the<br><a =
href=3D"mailto:ietf@ietf.org">ietf@ietf.org</a> mailing lists by =
2009-12-22. Exceptionally, <br>comments may be sent to <a =
href=3D"mailto:iesg@ietf.org">iesg@ietf.org</a> instead. In either case, =
please <br>retain the beginning of the Subject line to allow automated =
sorting.<br><br>The IPR disclosure at <br><a =
href=3D"https://datatracker.ietf.org/ipr/search/?option=3Ddocument_search&=
amp;id_document_tag=3D10128">https://datatracker.ietf.org/ipr/search/?opti=
on=3Ddocument_search&amp;id_document_tag=3D10128</a><br>was received =
after the working group requested publication. Please consider =
that<br>disclosure when preparing comments.<br><br>The file can be =
obtained =
via<br>http://www.ietf.org/internet-drafts/draft-ietf-sipping-config-frame=
work-16.txt<br><br><br>IESG discussion can be tracked =
via<br>https://datatracker.ietf.org/public/pidtracker.cgi?command=3Dview_i=
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________________<br>Sipping mailing list =
&nbsp;https://www.ietf.org/mailman/listinfo/sipping<br>This list is for =
NEW development of the application of SIP<br>Use =
sip-implementors@cs.columbia.edu for questions on current sip<br>Use =
sip@ietf.org for new developments of core =
SIP<br></div></blockquote></div><br></body></html>=

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From rjsparks@nostrum.com  Wed Dec  9 11:27:40 2009
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Subject: Re: [sipcore] draft-ietf-sipcore-sec-flows-01: document scope and intent
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On Nov 24, 2009, at 3:25 PM, Brian Hibbard wrote:

> A new version of the SIP Sec Flows draft is now available at http://www.ietf.org/internet-drafts/draft-ietf-sipcore-sec-flows-01.txt 
> .
>
>
> The feedback on the draft so far raises some interesting points  
> about the scope of the Sec Flows draft.  There are some basic issues  
> that need to be resolved for the draft to proceed.  Responses to  
> some of these issues may indicate the need for additional  
> document(s) or a change in scope of the draft.
>
>
> == Informative vs Normative ==
>
> The draft states in the first line of the introduction that it is  
> not normative for SIP.  That raises the question as to whether this  
> should be on the standards track.  Should this be informational or  
> BCP?

I don't remember why this was PS. Informational seems right. BCP is not.
>
>
> == Tutorial or Reference ==
>
> Is the draft intended to be a hitchhikers' guide to security for SIP  
> implementers?
I don't believe so - it was a set of examples.
>
> Does the draft intend to be a training tool to educate implementers  
> in PKI?
No.
> How far do we want to go in teaching developers how to generate and  
> manage certificates?
Only to the point that it's needed in order to explain what's  
happening in the examples.
> For example, are we trying to teach how to administer a Certificate  
> Authority and want to deal with how certs are generated in the real  
> world?
Very much no. Is something in the draft looking like it does that to  
you now?
> Or are we only concerned with the cryptography of the messages, and  
> we are only providing certificates and
>
> Are we only trying to provide specific reference examples for  
> helping verify implementations?
The last.
>
> If so, should we be providing test vectors instead?
I think you're focusing only on the crypto generation part of the  
draft (it is the hard part you jumped in to help with), but the examples
frame when what type of security mechanisms get used.
>  Is the document intending to recommend how things should be done,  
> how they might be done, or how they must be done?
The document provides some example flows. It is not intended to be  
comprehensive. In some cases it shows the only way
to do a certain thing. In others, it chooses from a range of options.

>
> == Document Title ==
>
> Depending on the resolution of the above, we may need to consider  
> changing the document's title.
I don't think we should be changing the document's scope to that  
degree at this time.



From fluffy@cisco.com  Wed Dec  9 11:55:33 2009
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To: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Subject: Re: [sipcore] Reviewing the sec-flows draft
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I get many emails from implementors that use this draft. I'm really =
would not wan to abandon it. The reason no one is very excited to work =
on it is that it has been pretty much done for many years now. I'm not =
even sure when I should bother to read it and I'm an author. If we are =
going to actually finish it, I'm glad to to get people to read it but =
the current process seems to just leave it in zombie land where no one =
does anything because there is no commitment one way or the other to do =
anything with it.

Cullen <with my individual contributor hat on>

On Dec 6, 2009, at 10:10 AM, Gonzalo Camarillo wrote:

> Folks,
>=20
> as you know, when we restructured the RAI area one of the plans was to =
avoid working on specifications people were not interested in any =
longer. Adam and I have had to assign a new editor (with cycles) to this =
document and have unsuccessfully asked for reviewers *many* times. If we =
do not see some activity related to this draft in the near future the =
only conclusion we can draw is that we should abandon it.
>=20
> So, if you think this is important, now it would be a good time for =
volunteering. Also, if you think we should abandon this draft, let us =
know as well.
>=20
> Thanks,
>=20
> Gonzalo
> SIPCORE co-chair
>=20
>=20
> Gonzalo Camarillo wrote:
>> Folks,
>> as you may have noticed, Brian has submitted a new revision of this =
draft and sent a couple of notes to the list (on November 24th) =
describing the topics on which he expects to get feedback. Please, send =
your comments to this list.
>> Thanks,
>> Gonzalo
>> SIPCORE co-chair
>> Gonzalo Camarillo wrote:
>>> Folks,
>>>=20
>>> as you know, we have asked for volunteers to review the sec-flows =
draft many times. We got a couple of reviews but we would like to get at =
least one dedicated review more. If you want to volunteer, please send =
your review to the list.
>>>=20
>>> Additionally, the reviews we got pointed out at issues that require =
WG consensus (as opposed to the editor simply going off and implementing =
them).
>>>=20
>>> As next steps, the draft's editor (Brian) will be submitting a new =
revision of the draft because the current one has expired. Then, he will =
be sending a set of emails to the list so that we can reach consensus on =
the issues that need discussion.
>>>=20
>>> Thanks,
>>>=20
>>> Gonzalo
>>> SIPCORE co-chair
>>> _______________________________________________
>>> sipcore mailing list
>>> sipcore@ietf.org
>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>=20
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


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Cc: SIPCORE <sipcore@ietf.org>, Brian Hibbard <brian@estacado.net>
Subject: Re: [sipcore] Reviewing the sec-flows draft
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Cullen Jennings wrote:
> I get many emails from implementors that use this draft.

Are those people using it to help them understand the specs?
Or in lieu of bothering to understand the specs?

Unfortunately I have the impression that there are a significant number 
of developers that implement from call flow documents and/or call traces 
of existing implementations rather than trying to understand the specs.

That is almost enough to motivate a "call flows considered harmful" paper.

I personally find call flows most useful:
1) by the writers of specs, to verify that the specs "work",
2) as a design output from implementors for review
    of the design prior to implementation
3) as use cases for planned new work

	Thanks,
	Paul

From christer.holmberg@ericsson.com  Wed Dec  9 12:57:15 2009
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Subject: Re: [sipcore] AD review draft-ietf-sipcore-199-00
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Hi,

I have now submitted version -02.

Regards,

Christer
=20

-----Original Message-----
From: Gonzalo Camarillo=20
Sent: 9. joulukuuta 2009 17:04
To: Christer Holmberg
Cc: Robert Sparks; sipcore@ietf.org;
draft-ietf-sipcore-199@tools.ietf.org; sipcore-chairs@tools.ietf.org
Subject: Re: [sipcore] AD review draft-ietf-sipcore-199-00

Hi Christer,

> Again, most changes were done (note the DTLS/SRTP issue, though), but=20
> in the wrong xml file. "Change done now" means that the changes will=20
> be in version -02.

why don't you submit 02 now so that people can follow the discussions
more clearly? You can submit 03 later if more changes are needed.

Thanks,

Gonzalo


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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           : Response Code for Indication of Terminated Dialog
	Author(s)       : C. Holmberg
	Filename        : draft-ietf-sipcore-199-02.txt
	Pages           : 16
	Date            : 2009-12-09

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

Status of this Memo

This Internet-Draft is submitted to IETF in full conformance with the
provisions of BCP 78 and BCP 79.

Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF), its areas, and its working groups.  Note that
other groups may also distribute working documents as Internet-
Drafts.

Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
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From fluffy@cisco.com  Wed Dec  9 22:04:51 2009
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On Dec 9, 2009, at 1:35 PM, Paul Kyzivat wrote:

>=20
>=20
> Cullen Jennings wrote:
>> I get many emails from implementors that use this draft.
>=20
> Are those people using it to help them understand the specs?
> Or in lieu of bothering to understand the specs?

Ouch, that is brutal. But uh, yah, some probably only read this spec =
because they could not find a wireshark capture that they could use to =
go implement from. But others are trying to use the stuff in here to =
test and to see at what stage their implementation went wrong.=20

>=20
> Unfortunately I have the impression that there are a significant =
number of developers that implement from call flow documents and/or call =
traces of existing implementations rather than trying to understand the =
specs.

uh yah, I see some call flows that certainly make me think some people =
do it that way

>=20
> That is almost enough to motivate a "call flows considered harmful" =
paper.
>=20
> I personally find call flows most useful:
> 1) by the writers of specs, to verify that the specs "work",
> 2) as a design output from implementors for review
>   of the design prior to implementation
> 3) as use cases for planned new work

Generally agree but in case of TLS or S/MIME, when you are implementing, =
it is really hard to figure out what went wrong. You just get a =
"security failed" error from underlying libraries and it is really nice =
to be able to compare your call flow to a "known good" one or even use =
some test vectors from it for testing.  Kumiko and I both did separate =
implementations of all this stuff then did messages between them in =
working on this draft.

>=20
> 	Thanks,
> 	Paul


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On Nov 24, 2009, at 2:25 PM, Brian Hibbard wrote:

> The draft states in the first line of the introduction that it is not =
normative for SIP.  That raises the question as to whether this should =
be on the standards track. Should this be informational or BCP?

I think we decided it need to be standards track when we added the =
normative statements about binary a long time ago. (I note that the =
"should" should be a "SHOULD".
=20=

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On Nov 24, 2009, at 2:26 PM, Brian Hibbard wrote:

> Add a few lines about RFC 3263. Add a few lines about how the UA =
decides to create a new TLS session or use an existing one (but not =
"connection-reuse" draft level of reuse complexity). Any volunteers for =
writing this?

I can write that if you don't fine some else to do it. How about:

When a UA goes to send a message to example.com, the UA can see if it =
already has a TLS connection to example.com and if it does, it MAY send =
the message over this connection. A UA SHOULD have some scheme for =
reusing connections as opening a new TLS connection for every message =
results in awful performance. Implementers are encouraged to read { add =
appropriate connection reuse refs here }.=20

Cullen (All my post on this draft are in my role as an individual =
contributor)


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On Nov 24, 2009, at 2:26 PM, Brian Hibbard wrote:

> =3D=3D Observed Interoperability Issues in section 6=3D=3D
> Who observed them? Is this experience from SIPits, etc? I think it =
would strengthen the idea that these are real-world, =
observed-in-the-wild issues to give sources.

Kumiko and I at SIPits are the main ref but this is all pretty uh =
fluffy, it's not like we have enough observation for any significant =
statistical sample. I have also seen S/MIME flows in operational =
networks but no one that uses those is going to be very interested in =
talking about them so I doubt we are going to get more input on this. =
I'm not wiling to put in an RFC names of vendors or product that had =
problems with their S/MIME flows - among other things that seems like it =
would violate the NDA and spirit of SIPit testing. Perhaps the best =
thing would be to remove the "problems observed" type text and just =
leave it at the  "If you are writing some code, make sure it does not =
have these problems" type text.=20






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On Nov 24, 2009, at 2:26 PM, Brian Hibbard wrote:

> There's some implicit stuff here that should be explicit. Do you mean =
to recommend never attaching certs? It's probably worth mentioning the =
message size limit issue. What do we mean by "it cannot be relied =
upon"--that we can't rely on the peer sending it, or it is unreliable =
when the peer does send it?

The first thing not the second. There is no guarantee that the peer will =
attach a cert to every message that would use it so you have to have =
some way to cache or get certs.=

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On Nov 24, 2009, at 2:26 PM, Brian Hibbard wrote:

> =46rom second bullet: What if all you've got is an IP address? Do we =
disallow IPAddress entries in SubjectAltName?

Good point - IPAddress are allowed in SubjectAltName AFAIC. I imagine =
that many people did not test this case and may have problems with it.=20=



From fluffy@cisco.com  Wed Dec  9 22:44:33 2009
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On Nov 24, 2009, at 2:26 PM, Brian Hibbard wrote:

> We should probably say something about CRLs. We need to get consensus =
on whether we want to recommend a method for retrieving CRLs.=20

I predict we will never get consensus on which way one should check CRLs =
thus I favor avoiding the CRL discussion but if others feel differently =
I don't care one way or the other.=20



From krisztian.kiss@nokia.com  Thu Dec 10 01:09:34 2009
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From: <krisztian.kiss@nokia.com>
To: <Martin.Thomson@andrew.com>, <salvatore.loreto@ericsson.com>
Date: Thu, 10 Dec 2009 10:08:36 +0100
Thread-Topic: [sipcore] Event rate control and GEOPRIV requirements
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Subject: Re: [sipcore] Event rate control and GEOPRIV requirements
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Hi Martin,

I will issue an updated version of the draft within the next couple of week=
s and adopt your preference making the Event header field optional for 2xx =
to NOTIFY (in order to cover the case when someone  doesn't need this parti=
cular feature but otherwise implementing the max-interval functionality).

Cheers,
Krisztian

-----Original Message-----
From: ext Thomson, Martin [mailto:Martin.Thomson@andrew.com]=20
Sent: Sunday, November 08, 2009 10:59 PM
To: Kiss Krisztian (Nokia-CIC/MtView); salvatore.loreto@ericsson.com
Cc: sipcore@ietf.org
Subject: RE: [sipcore] Event rate control and GEOPRIV requirements

I don't think that there is a conflict.

If the subscriber wishes to change details, then they can provide the heade=
r.  If they do not provide the header - as is the existing behaviour - the =
values do not change.

Of course, if the _header_ is present, then the parameter must be, but that=
 is not what the text in the document says.

Making this mandatory also makes it mandatory for those who implement this =
feature but don't intend to use it.  Let's not break existing event package=
 implementations.

--Martin

> -----Original Message-----
> From: krisztian.kiss@nokia.com [mailto:krisztian.kiss@nokia.com]
> Sent: Monday, 9 November 2009 3:50 PM
> To: salvatore.loreto@ericsson.com; Thomson, Martin
> Cc: sipcore@ietf.org
> Subject: RE: [sipcore] Event rate control and GEOPRIV requirements
>=20
> Hi,
>=20
> Making it optional may create a conflict with the following statement:
>=20
>    A subscriber that wishes to remove the minimum rate control from
>    notifications MUST indicate so by not including a "max-interval"
>    Event header field parameter in a subsequent SUBSCRIBE request or a
>    200-class response to the NOTIFY request.
>=20
> An Event header field without the "max-interval" parameter means that=20
> the subscriber is deleting the "max-interval" feature from=20
> notifications. Should we now allocate a special meaning for the case=20
> when no Event header field exists in the 2xx response to the NOTIFY?
> Does that mean the "max-interval" parameter is not touched from the=20
> previous value? Or something else? I'd prefer the same behavior as for=20
> the SUBSCRIBE request, i.e. the "max-interval" parameter is always=20
> present reflecting the current value and the lack of the parameter=20
> means that the feature is deleted from notifications.
>=20
> Existing implementations do not support Event header field in the 2xx=20
> responses to NOTIFY, however new implementations adding the rate-=20
> control feature to notifications will support adding the rate control=20
> parameters to the Event header field in SUBSCRIBE and 2xx to NOTIFY as=20
> well as the Subscription-State header field in NOTIFYs. As RFC 3265=20
> did not define the Event header field to 2xx to NOTIFY,=20
> draft-ietf-sipcore- event-rate-control has to do this in order to have=20
> a placeholder for the new parameter.
>=20
> Cheers,
> Krisztian
>=20
> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On=20
> Behalf Of ext Salvatore Loreto
> Sent: Sunday, November 08, 2009 9:49 PM
> To: Thomson, Martin
> Cc: sipcore@ietf.org
> Subject: Re: [sipcore] Event rate control and GEOPRIV requirements
>=20
> Hi Martin,
>=20
> good point, I think we can relax it to optional .
>=20
> /Sal
>=20
> Thomson, Martin wrote:
> > Hi Sal,
> >
> > Thanks for the pointer.
> >
> > Regarding section 8.3, which addresses the requirement in question:
> >
> > I'm not sure why you would make the Event header mandatory in the=20
> > 2xx
> responses.  Optional seems to make more sense to me.  Existing=20
> implementations don't do this, after all.
> >
> >
> >> -----Original Message-----
> >> From: Salvatore Loreto [mailto:salvatore.loreto@ericsson.com]
> >> Sent: Monday, 9 November 2009 2:25 PM
> >> To: Thomson, Martin
> >> Cc: sipcore@ietf.org
> >> Subject: Re: [sipcore] Event rate control and GEOPRIV requirements
> >>
> >> Hi Martin,
> >>
> >> an updated version of the event-rate-control draft was submitted=20
> >> two days after the IETF76 deadline, for this reason it is not yet=20
> >> on the I-D repository, however it will be available soon
> >>
> >> for the time being you can fetch a copy from
> >> http://krkiss.letolt.info/draft-ietf-sipcore-event-rate-control-
> 01.tx
> >> t
> >>
> >> /Sal
> >>
> >> Thomson, Martin wrote:
> >>
> >>> Is the -00 version the latest issue of this draft?  I expected a -
> 01
> >>>
> >> and so was under the impression that we were discussion changes=20
> >> already made.
> >>
> >>> The following two minor changes were requested out of the GEOPRIV=20
> >>> use
> >>>
> >> cases:
> >>
> >>>  - note that the rate control parameters can be updated in=20
> >>> response
> >>>
> >> messages (i.e. the 200 response to an initial NOTIFY); the notifier=20
> >> then indicates the agreed (and possibility modified) values in=20
> >> subsequent NOTIFYs
> >>
> >>>  - note that rate control parameters can be changed at any time by
> >>>
> >> the notifier, based on policy- or implementation-determined=20
> >> constraints
> >>
> >>> This is what we discussed in the meeting.
> >>>
> >>> I think that the second one is OK in the version -00 draft.  I
> could
> >>>
> >> not find anywhere that mentioned updating event parameters in a=20
> >> response message.
> >>
> >>> --Martin
> >>>
> >>> _______________________________________________
> >>> sipcore mailing list
> >>> sipcore@ietf.org
> >>> https://www.ietf.org/mailman/listinfo/sipcore
> >>>
> >>>
> >
> >
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From gonzalo.camarillo@ericsson.com  Thu Dec 10 03:39:21 2009
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Subject: [sipcore] WGLC draft-ietf-sipcore-info-events-03.txt
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Folks,

Christer has submitted a new version of the INFO Events draft (see email 
below) that is supposed to address all the comments received so far. 
Since there have been extensive discussions on this topic since the last 
time we WGLCed this draft, we are going to WGLC it again.

http://tools.ietf.org/html/draft-ietf-sipcore-info-events-03

So, please, have a look at this new version and let us know whether or 
not you also think all comments have been addressed. Of course, you are 
also welcome to submit new comments should you have some. This WGLC will 
end on December 24th.

Thanks,

Gonzalo
SIPCORE co-chair

-------- Original Message --------
Subject: 	[sipcore] Draft new version: draft-ietf-sipcore-info-events
Date: 	Wed, 2 Dec 2009 09:59:24 +0100
From: 	Christer Holmberg <christer.holmberg@ericsson.com>
To: 	<sipcore@ietf.org>




Hi,

I've submitted a new version of the info draft. Hopefully all comments
and feedback have now been addressed.

The biggest changes are:

- Section 4.2: The usage of the Recv-Info header has been clarified,
some simple rules have been added, and there were some discussions that
since the re-INVITE fallback issue also affects Recv-Info (if you
receive a Recv-Info in a 18x to a re-INVITE, and the re-INVITE then
fails) and that we need some words about that. The text now says that IF
the request contains a Recv-Info header, the response must also contain it.

- Section 11.4: Contains new wording about the IANA process for
registering info packages, including the expert review etc.

- Section 12: Additional examples have been added


Please note that I did NOT remove Recv-Info in PRACKs, since there in my
opinion was no concensus to do so.


The draft can also be found at:

_http://users.piuha.net/cholmber/drafts/draft-ietf-sipcore-info-events-03.txt_ 




Regards,

Christer



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_______________________________________________
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Subject: Re: [sipcore] Reviewing the sec-flows draft
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Hi,

thanks all of you who replied showing interest in this work and, more 
importantly, started responding to the questions Brian sent some time 
ago. Let's make sure we keep the ball rolling now.

Thanks,

Gonzalo

Gonzalo Camarillo wrote:
> Folks,
> 
> as you know, when we restructured the RAI area one of the plans was to 
> avoid working on specifications people were not interested in any 
> longer. Adam and I have had to assign a new editor (with cycles) to this 
> document and have unsuccessfully asked for reviewers *many* times. If we 
> do not see some activity related to this draft in the near future the 
> only conclusion we can draw is that we should abandon it.
> 
> So, if you think this is important, now it would be a good time for 
> volunteering. Also, if you think we should abandon this draft, let us 
> know as well.
> 
> Thanks,
> 
> Gonzalo
> SIPCORE co-chair
> 
> 
> Gonzalo Camarillo wrote:
>> Folks,
>>
>> as you may have noticed, Brian has submitted a new revision of this 
>> draft and sent a couple of notes to the list (on November 24th) 
>> describing the topics on which he expects to get feedback. Please, 
>> send your comments to this list.
>>
>> Thanks,
>>
>> Gonzalo
>> SIPCORE co-chair
>>
>> Gonzalo Camarillo wrote:
>>> Folks,
>>>
>>> as you know, we have asked for volunteers to review the sec-flows 
>>> draft many times. We got a couple of reviews but we would like to get 
>>> at least one dedicated review more. If you want to volunteer, please 
>>> send your review to the list.
>>>
>>> Additionally, the reviews we got pointed out at issues that require 
>>> WG consensus (as opposed to the editor simply going off and 
>>> implementing them).
>>>
>>> As next steps, the draft's editor (Brian) will be submitting a new 
>>> revision of the draft because the current one has expired. Then, he 
>>> will be sending a set of emails to the list so that we can reach 
>>> consensus on the issues that need discussion.
>>>
>>> Thanks,
>>>
>>> Gonzalo
>>> SIPCORE co-chair
>>> _______________________________________________
>>> 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
> 


From vkg@alcatel-lucent.com  Thu Dec 10 06:52:30 2009
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Subject: Re: [sipcore] draft-ietf-sipcore-sec-flows-01: technical issues
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Cullen Jennings wrote:
> On Nov 24, 2009, at 2:26 PM, Brian Hibbard wrote:
> 
>> Add a few lines about RFC 3263. Add a few lines about how the UA
>> decides to create a new TLS session or use an existing one (but not
>> "connection-reuse" draft level of reuse complexity). Any volunteers
>> for writing this?
> 
> I can write that if you don't fine some else to do it. How about:
> 
> When a UA goes to send a message to example.com, the UA can see if it
> already has a TLS connection to example.com and if it does, it MAY
> send the message over this connection. A UA SHOULD have some scheme
> for reusing connections as opening a new TLS connection for every
> message results in awful performance. Implementers are encouraged to
> read { add appropriate connection reuse refs here }.

This is exactly what connect-reuse says, except of course, in
much more detail about how the UA goes on to determine "if it
already has a TLS connection to example.com".  I don't mind
putting a capsule summary of the above form in sec-flows, but
we should follow it up with a reference to connect-reuse.

Thanks,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
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From vkg@alcatel-lucent.com  Thu Dec 10 07:03:21 2009
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Subject: Re: [sipcore] draft-ietf-sipcore-sec-flows-01: technical issues
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Cullen Jennings wrote:
> On Nov 24, 2009, at 2:26 PM, Brian Hibbard wrote:
> 
>> From second bullet: What if all you've got is an IP address? Do we
>> disallow IPAddress entries in SubjectAltName?
> 
> Good point - IPAddress are allowed in SubjectAltName AFAIC. I imagine
> that many people did not test this case and may have problems with
> it.

We can add it as a test case in Section 8.  Something along the
lines of:

   12. TLS : Good peer certificate (CN of Subject empty, and
       subjectAltName extension contains an iPAddress stored
       in the octet string in network byte order form as
       specified in [RFC791].)

Thanks,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
Web:   http://ect.bell-labs.com/who/vkg/

From adam@nostrum.com  Thu Dec 10 07:38:23 2009
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Call for consensus: adopting	draft-camarillo-sipcore-reinvite
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[as chair]

Hearing significant support, and no objections, I declare consensus in 
favor of adopting this document. I have instructed the authors to submit 
a new version as draft-ietf-sipcore-reinvite.

Thank you.

/a

On 12/4/09 5:49 PM, Adam Roach wrote:
> [as chair]
>
> Regarding the following milestone on our charter:
>
>   Invite Transaction Handling Correction to IESG (PS)
>
> To fulfill this milestone, there are two potential problems that may 
> need to be addressed. One of them relates to handling of 200-class 
> responses to INVITE messages; the other relates to re-INVITE and 
> target refresh handling.
>
> The first problem is addressed by an adopted working group item 
> (draft-ietf-sipcore-invfix). We do not currently have an adopted item 
> for the second problem.
>
> The chairs would like to solicit feedback from the group regarding 
> adoption of draft-camarillo-sipcore-reinvite as a working group item 
> to fix the second set of problems. Please comment on the SIPCORE 
> mailing list. If you agree with this adoption, a simple note stating 
> as much is encouraged: we cannot determine consensus without feedback, 
> and silence is not very informative.
>
> Thanks!
>
> /a
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From gonzalo.camarillo@ericsson.com  Thu Dec 10 12:57:02 2009
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] Call for consensus:	adopting	draft-camarillo-sipcore-reinvite
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Hi,

I have just submitted the draft:
http://www.ietf.org/id/draft-ietf-sipcore-reinvite-00.txt

The only thing I have changed in this revision is the file name.

Cheers,

Gonzalo

Adam Roach wrote:
> [as chair]
> 
> Hearing significant support, and no objections, I declare consensus in 
> favor of adopting this document. I have instructed the authors to submit 
> a new version as draft-ietf-sipcore-reinvite.
> 
> Thank you.
> 
> /a
> 
> On 12/4/09 5:49 PM, Adam Roach wrote:
>> [as chair]
>>
>> Regarding the following milestone on our charter:
>>
>>   Invite Transaction Handling Correction to IESG (PS)
>>
>> To fulfill this milestone, there are two potential problems that may 
>> need to be addressed. One of them relates to handling of 200-class 
>> responses to INVITE messages; the other relates to re-INVITE and 
>> target refresh handling.
>>
>> The first problem is addressed by an adopted working group item 
>> (draft-ietf-sipcore-invfix). We do not currently have an adopted item 
>> for the second problem.
>>
>> The chairs would like to solicit feedback from the group regarding 
>> adoption of draft-camarillo-sipcore-reinvite as a working group item 
>> to fix the second set of problems. Please comment on the SIPCORE 
>> mailing list. If you agree with this adoption, a simple note stating 
>> as much is encouraged: we cannot determine consensus without feedback, 
>> and silence is not very informative.
>>
>> Thanks!
>>
>> /a
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
> 
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 


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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           : Re-INVITE and Target-refresh Request Handling in the Session Initiation Protocol (SIP)
	Author(s)       : G. Camarillo, et al.
	Filename        : draft-ietf-sipcore-reinvite-00.txt
	Pages           : 22
	Date            : 2009-12-10

In this document, we clarify the handling of re-INVITEs in SIP.  We
clarify in which situations a UAS (User Agent Server) should generate
a success response and in which situations a UAS should generate an
error response to a re-INVITE.  Additionally, we clarify issues
related to target-refresh requests.

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From fluffy@cisco.com  Thu Dec 10 20:05:12 2009
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To: Harhit Tam <harhittam@gmail.com>
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Subject: Re: [sipcore] e2e sip question
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Yes you can. People sometimes set up SIP PSTN gateways bewtween a few =
PBX more or less like this. It's also pretty common between a PBX and =
voicemail system or conferencing server.=20

On Nov 20, 2009, at 8:04 AM, Harhit Tam wrote:

> Hi all,=20
>=20
> Can we use e2e SIP (without proxies) using only the destination's DNS =
name?
>=20
> That is Bob's phone has a DNS name bob.domain.org, and Alice would =
like
> to make a call to that DNS name directly without using a SIP URI in =
the form
> of an e-mail address.
>=20
> How can we use SIP in this way?
>=20
> Thank you in advance.=20
>=20
> Regards.
>=20
> Harhit
>=20
> =20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From drage@alcatel-lucent.com  Fri Dec 11 06:08:29 2009
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From: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
To: Cullen Jennings <fluffy@cisco.com>, Brian Hibbard <brian@estacado.net>
Date: Fri, 11 Dec 2009 15:07:43 +0100
Thread-Topic: [sipcore] draft-ietf-sipcore-sec-flows-01: document scope and intent
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Cc: SIPCORE <sipcore@ietf.org>, Kumiko Ono <kumiko@cs.columbia.edu>
Subject: Re: [sipcore] draft-ietf-sipcore-sec-flows-01: document scope and	intent
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If so, you arbitrarily decided to change it because as far as I know it was=
 informational during its complete lifetime in the SIP group, and I remembe=
r no discussion in the transfer to SIPCORE that made it standards track. Ce=
rtainly my final SIP status slides (and the ones before and the ones before=
 that ...) also say "Candidate: Informational". The first version that carr=
ies an "intended status" line is the 1st sipcore version, and this is at va=
riance with the charter pages.

The last SIP charter pages clearly say:
Mar 2009    Example security flows to WGLC (Informational)=20
Jul 2009    Example security flows to IESG (Informational) =20

And the SIPCORE charter pages state:

Oct 2009    Example security flows to IESG (Informational)=20

And from my own perspective this should still be informational. If you feel=
 there is standards track information in here, then pull it out and submit =
it as a separate draft, and then the WG can decide whether they want to pro=
ceed with such a standards track document.

I'd also note to all authors that if you receive private messages of intere=
st, then at least please let the WG chairs know that you have had those exp=
ressions of interest. If you received any of those messages on sec-flows du=
ring its SIP lifetime, the ex SIP WG chairs certainly have no evidence of t=
hem ever having been received - and that is essential information to the PR=
OTO writeups. I assume the SIPCORE chairs are now in the same boat, having =
no email evidence because a small set of exchanges in the past on SIP and S=
IPCORE that anyone is even interested in this document.

regards

Keith

> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Cullen Jennings
> Sent: Thursday, December 10, 2009 6:14 AM
> To: Brian Hibbard
> Cc: SIPCORE; Kumiko Ono
> Subject: Re: [sipcore] draft-ietf-sipcore-sec-flows-01:=20
> document scope and intent
>=20
>=20
> On Nov 24, 2009, at 2:25 PM, Brian Hibbard wrote:
>=20
> > The draft states in the first line of the introduction that=20
> it is not normative for SIP.  That raises the question as to=20
> whether this should be on the standards track. Should this be=20
> informational or BCP?
>=20
> I think we decided it need to be standards track when we=20
> added the normative statements about binary a long time ago.=20
> (I note that the "should" should be a "SHOULD".
> =20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =

From drage@alcatel-lucent.com  Fri Dec 11 06:09:17 2009
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From: "DRAGE, Keith (Keith)" <drage@alcatel-lucent.com>
To: Cullen Jennings <fluffy@cisco.com>, Brian Hibbard <brian@estacado.net>
Date: Fri, 11 Dec 2009 15:07:55 +0100
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Cc: SIPCORE <sipcore@ietf.org>, Kumiko Ono <kumiko@cs.columbia.edu>
Subject: Re: [sipcore] draft-ietf-sipcore-sec-flows-01: technical issues
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NO.

In my view this is an informational document. Description only in the text.

See previous message.=20

Keith

> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Cullen Jennings
> Sent: Thursday, December 10, 2009 6:28 AM
> To: Brian Hibbard
> Cc: SIPCORE; Kumiko Ono
> Subject: Re: [sipcore] draft-ietf-sipcore-sec-flows-01:=20
> technical issues
>=20
>=20
> On Nov 24, 2009, at 2:26 PM, Brian Hibbard wrote:
>=20
> > Add a few lines about RFC 3263. Add a few lines about how=20
> the UA decides to create a new TLS session or use an existing=20
> one (but not "connection-reuse" draft level of reuse=20
> complexity). Any volunteers for writing this?
>=20
> I can write that if you don't fine some else to do it. How about:
>=20
> When a UA goes to send a message to example.com, the UA can=20
> see if it already has a TLS connection to example.com and if=20
> it does, it MAY send the message over this connection. A UA=20
> SHOULD have some scheme for reusing connections as opening a=20
> new TLS connection for every message results in awful=20
> performance. Implementers are encouraged to read { add=20
> appropriate connection reuse refs here }.=20
>=20
> Cullen (All my post on this draft are in my role as an=20
> individual contributor)
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =

From ben@nostrum.com  Fri Dec 11 09:06:00 2009
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Subject: Re: [sipcore] draft-ietf-sipcore-sec-flows-01: technical issues
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On reflection, I agree this draft should not contain normative =
statements for this sort of thing.  If we run across a real need for =
such language in this draft, perhaps it means we have other standards =
work to do.

In this case, is there nothing in RAI's entire body of work that talks =
about this normatively?



On Dec 11, 2009, at 8:07 AM, DRAGE, Keith (Keith) wrote:

> NO.
>=20
> In my view this is an informational document. Description only in the =
text.
>=20
> See previous message.=20
>=20
> Keith
>=20
>> -----Original Message-----
>> From: sipcore-bounces@ietf.org=20
>> [mailto:sipcore-bounces@ietf.org] On Behalf Of Cullen Jennings
>> Sent: Thursday, December 10, 2009 6:28 AM
>> To: Brian Hibbard
>> Cc: SIPCORE; Kumiko Ono
>> Subject: Re: [sipcore] draft-ietf-sipcore-sec-flows-01:=20
>> technical issues
>>=20
>>=20
>> On Nov 24, 2009, at 2:26 PM, Brian Hibbard wrote:
>>=20
>>> Add a few lines about RFC 3263. Add a few lines about how=20
>> the UA decides to create a new TLS session or use an existing=20
>> one (but not "connection-reuse" draft level of reuse=20
>> complexity). Any volunteers for writing this?
>>=20
>> I can write that if you don't fine some else to do it. How about:
>>=20
>> When a UA goes to send a message to example.com, the UA can=20
>> see if it already has a TLS connection to example.com and if=20
>> it does, it MAY send the message over this connection. A UA=20
>> SHOULD have some scheme for reusing connections as opening a=20
>> new TLS connection for every message results in awful=20
>> performance. Implementers are encouraged to read { add=20
>> appropriate connection reuse refs here }.=20
>>=20
>> Cullen (All my post on this draft are in my role as an=20
>> individual contributor)
>>=20
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
>>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From RIFATYU@nortel.com  Sat Dec 12 10:54:36 2009
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From: "Rifaat Shekh-Yusef" <rifatyu@nortel.com>
To: "Christer Holmberg" <christer.holmberg@ericsson.com>, <sipcore@ietf.org>
Subject: Re: [sipcore] Draft new version: keep
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Hi Christer,
=20
I read your document (sipcore-keep-01) and I have the following
questions (I am sorry if these questions were asked before):
=20
RFC 5626 recommends the STUN Keep-Alive technique with connection-less
transports and CRLF Keep-Alive technique with connection-oriented
transports.
Is this document recommending the client should use STUN Keep-Alive
technique for both connection-oriented and connection-less transports?
Why did you choose STUN over CRLF?
=20
The header of Section 8 implies that there is an overlap between this
document and the connect-reuse draft, but when you read the section you
realize that this is not the case.
Why do you think that this section is needed? Does it matter if the
connection between the UA and the proxy uses one TCP connection or two
TCP connections?
=20
Thanks,
 Rifaat
=20


________________________________

From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
Behalf Of Christer Holmberg
Sent: Tuesday, December 01, 2009 6:18 AM
To: sipcore@ietf.org
Subject: [sipcore] Draft new version: keep





Hi,=20

I've submitted a new version of the keep draft. There are no changes
from the previous version, but it had expired.=20

I will produce a new version, with new content, once we get the info
issues resolved - it has really taken more time than expected...=20

A new version of the info draft will be submitted tomorrow.=20

Regards,=20

Christer=20


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>Draft new version: keep</TITLE>
<META content=3D"text/html; charset=3Dus-ascii" =
http-equiv=3DContent-Type>
<META name=3DGENERATOR content=3D"MSHTML 8.00.6001.18854"></HEAD>
<BODY>
<DIV><SPAN class=3D108062318-12122009><FONT color=3D#0000ff size=3D2 =
face=3DArial>Hi=20
Christer,</FONT></SPAN></DIV>
<DIV><SPAN class=3D108062318-12122009><FONT color=3D#0000ff size=3D2=20
face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D108062318-12122009><FONT color=3D#0000ff size=3D2 =
face=3DArial>I read=20
your document (sipcore-keep-01)&nbsp;and I have the following questions =
(I am=20
sorry if these questions were asked before):</FONT></SPAN></DIV>
<DIV><SPAN class=3D108062318-12122009><FONT color=3D#0000ff size=3D2=20
face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D108062318-12122009><FONT color=3D#0000ff size=3D2 =
face=3DArial>RFC=20
5626 recommends the STUN Keep-Alive technique with connection-less =
transports=20
and CRLF Keep-Alive technique with connection-oriented=20
transports.</FONT></SPAN></DIV>
<DIV><SPAN class=3D108062318-12122009><FONT color=3D#0000ff size=3D2 =
face=3DArial>Is=20
this document recommending the&nbsp;client should use STUN Keep-Alive =
technique=20
for both connection-oriented and connection-less =
transports?</FONT></SPAN></DIV>
<DIV><SPAN class=3D108062318-12122009><FONT color=3D#0000ff size=3D2 =
face=3DArial>Why=20
did you choose STUN over CRLF?</FONT></SPAN></DIV>
<DIV><SPAN class=3D108062318-12122009><FONT color=3D#0000ff size=3D2=20
face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D108062318-12122009><FONT color=3D#0000ff size=3D2 =
face=3DArial>The=20
header of Section 8 implies that there is an overlap between this =
document and=20
the connect-reuse draft, but when you read the section you realize that =
this is=20
not the case.</FONT></SPAN></DIV>
<DIV><SPAN class=3D108062318-12122009><FONT color=3D#0000ff size=3D2 =
face=3DArial>Why do=20
you think that this section is needed?&nbsp;Does it matter if the =
connection=20
between the UA&nbsp;and the proxy&nbsp;uses one TCP connection or two =
TCP=20
connections?</FONT></SPAN></DIV>
<DIV><SPAN class=3D108062318-12122009><FONT color=3D#0000ff size=3D2=20
face=3DArial></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D108062318-12122009><FONT color=3D#0000ff size=3D2=20
face=3DArial>Thanks,</FONT></SPAN></DIV>
<DIV><SPAN class=3D108062318-12122009><FONT color=3D#0000ff size=3D2=20
face=3DArial>&nbsp;Rifaat</FONT></SPAN></DIV>
<DIV><SPAN class=3D108062318-12122009><FONT color=3D#0000ff size=3D2=20
face=3DArial></FONT></SPAN>&nbsp;</DIV><FONT color=3D#0000ff size=3D2=20
face=3DArial></FONT><FONT color=3D#0000ff size=3D2 =
face=3DArial></FONT><BR>
<DIV dir=3Dltr lang=3Den-us class=3DOutlookMessageHeader align=3Dleft>
<HR tabIndex=3D-1>
<FONT size=3D2 face=3DTahoma><B>From:</B> sipcore-bounces@ietf.org=20
[mailto:sipcore-bounces@ietf.org] <B>On Behalf Of </B>Christer=20
Holmberg<BR><B>Sent:</B> Tuesday, December 01, 2009 6:18 =
AM<BR><B>To:</B>=20
sipcore@ietf.org<BR><B>Subject:</B> [sipcore] Draft new version:=20
keep<BR></FONT><BR></DIV>
<DIV></DIV><!-- Converted from text/rtf format --><FONT color=3D#0000ff =
size=3D2=20
face=3DArial></FONT><FONT color=3D#0000ff size=3D2 =
face=3DArial></FONT><BR>
<P><FONT size=3D2 face=3DArial>Hi,</FONT> </P>
<P><FONT size=3D2 face=3DArial>I've submitted a new version of the keep =
draft. There=20
are no changes from the previous version, but it had expired.</FONT> =
</P>
<P><FONT size=3D2 face=3DArial>I will produce a new version, with new =
content, once=20
we get the info issues resolved - it has really taken more time than =
expected...=20
</FONT></P>
<P><FONT size=3D2 face=3DArial>A new version of the info draft will be =
submitted=20
tomorrow.</FONT> </P>
<P><FONT size=3D2 face=3DArial>Regards,</FONT> </P>
<P><FONT size=3D2 face=3DArial>Christer</FONT> </P></BODY></HTML>

------_=_NextPart_001_01CA7B5C.77FDC088--

From gunnar.hellstrom@omnitor.se  Sat Dec 12 14:35:49 2009
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Subject: Re: [sipcore] I-D ACTION:draft-ietf-sipcore-location-conveyance-01.txt
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 I think it would be useful if draft-ietf-sipcore-location-conveyance moved
on in status.

/Gunnar
-----Original Message-----
From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-bounces@ietf.org]
On Behalf Of Internet-Drafts@ietf.org
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To: i-d-announce@ietf.org
Cc: sipcore@ietf.org
Subject: I-D ACTION:draft-ietf-sipcore-location-conveyance-01.txt 

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

	Title		: Location Conveyance for the Session Initiation
Protocol
	Author(s)	: J. Polk, B. Rosen
	Filename	: draft-ietf-sipcore-location-conveyance-01.txt
	Pages		: 48
	Date		: 2009-7-13
	
This document defines an extension to the Session Initiation 
   Protocol (SIP) to convey geographic location information from one 
   SIP entity to another SIP entity.  The extension covers end-to-end 
   conveyance as well as location-based routing, where SIP servers 
   make routing decisions based on the location of the user agent 
   client.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sipcore-location-conveyance-0
1.txt

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From christer.holmberg@ericsson.com  Sun Dec 13 23:19:52 2009
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Rifaat Shekh-Yusef' <rifatyu@nortel.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Mon, 14 Dec 2009 08:18:15 +0100
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Hi,
 =20
>I read your document (sipcore-keep-01) and I have the following questions =
(I am sorry if these questions were asked before):
>=20
>RFC 5626 recommends the STUN Keep-Alive technique with connection-less tra=
nsports and CRLF Keep-Alive technique with connection-oriented transports.
>Is this document recommending the client should use STUN Keep-Alive techni=
que for both connection-oriented and connection-less transports?

No.

The idea is to use the keep-alive mechanism defined in RFC 5626.

The purpose of the draft is only to allow entities to negotiate the usage o=
f keep-alive, if they don't support outbound.

>Why did you choose STUN over CRLF?
>
>The header of Section 8 implies that there is an overlap between this docu=
ment and the connect-reuse draft, but when you read the section you realize=
=20
>that this is not the case.
>Why do you think that this section is needed? Does it matter if the connec=
tion between the UA and the proxy uses one TCP connection or two TCP=20
>connections?

At one point I was asked to add that section, just to point out the differe=
nces between keep-alive and connect-reuse.

Regards,

Christer




=20


________________________________

From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf =
Of Christer Holmberg
Sent: Tuesday, December 01, 2009 6:18 AM
To: sipcore@ietf.org
Subject: [sipcore] Draft new version: keep





Hi,=20

I've submitted a new version of the keep draft. There are no changes from t=
he previous version, but it had expired.=20

I will produce a new version, with new content, once we get the info issues=
 resolved - it has really taken more time than expected...=20

A new version of the info draft will be submitted tomorrow.=20

Regards,=20

Christer=20


From RIFATYU@nortel.com  Mon Dec 14 07:48:11 2009
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From: "Rifaat Shekh-Yusef" <rifatyu@nortel.com>
To: "Christer Holmberg" <christer.holmberg@ericsson.com>, <sipcore@ietf.org>
Subject: Re: [sipcore] Draft new version: keep
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> =20
>>I read your document (sipcore-keep-01) and I have the following
questions (I am sorry if these questions were asked before):
>>=20
>>RFC 5626 recommends the STUN Keep-Alive technique with connection-less
transports and CRLF Keep-Alive technique with connection-oriented
transports.
>>Is this document recommending the client should use STUN Keep-Alive
technique for both connection-oriented and connection-less transports?
>
>No.
>
>The idea is to use the keep-alive mechanism defined in RFC 5626.
>
>The purpose of the draft is only to allow entities to negotiate the
usage of keep-alive, if they don't support outbound.

Then section 5 should be clarified to reflect this, as this is not clear
from the current text.


>>Why did you choose STUN over CRLF?
>>
>>The header of Section 8 implies that there is an overlap between this=20
>>document and the connect-reuse draft, but when you read the section
you realize that this is not the case.
>>Why do you think that this section is needed? Does it matter if the=20
>>connection between the UA and the proxy uses one TCP connection or two
TCP connections?
>
>At one point I was asked to add that section, just to point out the
differences between keep-alive and connect-reuse.

This section does not seem to add any value to this document.


Regards,
 Rifaat

=20


________________________________

From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
Behalf Of Christer Holmberg
Sent: Tuesday, December 01, 2009 6:18 AM
To: sipcore@ietf.org
Subject: [sipcore] Draft new version: keep





Hi,=20

I've submitted a new version of the keep draft. There are no changes
from the previous version, but it had expired.=20

I will produce a new version, with new content, once we get the info
issues resolved - it has really taken more time than expected...=20

A new version of the info draft will be submitted tomorrow.=20

Regards,=20

Christer=20


From christer.holmberg@ericsson.com  Mon Dec 14 23:05:19 2009
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Rifaat Shekh-Yusef' <rifatyu@nortel.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Tue, 15 Dec 2009 08:05:02 +0100
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Hi,=20

>>>I read your document (sipcore-keep-01) and I have the following
>>>questions (I am sorry if these questions were asked before):
>>>RFC 5626 recommends the STUN Keep-Alive technique with connection-less
>>>transports and CRLF Keep-Alive technique with connection-oriented transp=
orts.
>>>Is this document recommending the client should use STUN Keep-Alive
>>>technique for both connection-oriented and connection-less transports?
>>
>>No.
>>
>>The idea is to use the keep-alive mechanism defined in RFC 5626.
>>
>>The purpose of the draft is only to allow entities to negotiate the
>>usage of keep-alive, if they don't support outbound.
>
>Then section 5 should be clarified to reflect this, as this is not clear f=
rom the current text.

Section 1 says:

"This specification defines a new SIP Via header parameter, "keep",which ca=
n be used by a UA to indicate support of the keep-alive
techniques defined in [I-D.ietf-sip-outbound]."

Section 5 also mentions, in a number of places, "keep-alive techniques defi=
ned in outbound", but I will take a look whether it can be more clear.

>>>Why did you choose STUN over CRLF?
>>>
>>>The header of Section 8 implies that there is an overlap between this=20
>>>document and the connect-reuse draft, but when you read the section
>>>you realize that this is not the case.
>>>Why do you think that this section is needed? Does it matter if the=20
>>>connection between the UA and the proxy uses one TCP connection or two T=
CP connections?
>>
>>At one point I was asked to add that section, just to point out the diffe=
rences between keep-alive and connect-reuse.
>
>This section does not seem to add any value to this document.

I would have to check the archieves for the discussions about it, but in ge=
neral I would not like to remove chapters which we have agreed to add.

I guess the people who wanted to chapter (Vijay?) could comment.

Regards,

Christer=

From RIFATYU@nortel.com  Tue Dec 15 06:22:54 2009
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From: "Rifaat Shekh-Yusef" <rifatyu@nortel.com>
To: "Christer Holmberg" <christer.holmberg@ericsson.com>, <sipcore@ietf.org>
Subject: Re: [sipcore] Draft new version: keep
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Hi Christer,

>>>>I read your document (sipcore-keep-01) and I have the following=20
>>>>questions (I am sorry if these questions were asked before):
>>>>RFC 5626 recommends the STUN Keep-Alive technique with=20
>>>>connection-less transports and CRLF Keep-Alive technique with
connection-oriented transports.
>>>>Is this document recommending the client should use STUN Keep-Alive=20
>>>>technique for both connection-oriented and connection-less
transports?
>>>
>>>No.
>>>
>>>The idea is to use the keep-alive mechanism defined in RFC 5626.
>>>
>>>The purpose of the draft is only to allow entities to negotiate the=20
>>>usage of keep-alive, if they don't support outbound.
>>
>>Then section 5 should be clarified to reflect this, as this is not
clear from the current text.
>
>Section 1 says:
>
>"This specification defines a new SIP Via header parameter,
"keep",which can be used by a UA to indicate support of the keep-alive
techniques defined in >[I-D.ietf-sip-outbound]."
>
>Section 5 also mentions, in a number of places, "keep-alive techniques
defined in outbound", but I will take a look whether it can be more
clear.

The very first paragraph in section 5 says the following:

   "A SIP entity aciting as a UAC which supports the method defined in
   this specification MUST act as a STUN client
   [I-D.ietf-behave-rfc3489bis], and MUST support the amount of STUN
   which is required to apply the STUN keep-alive technique
   [I-D.ietf-sip-outbound]."

The way I understand this paragraph and the use of "MUST" is that you
are mandating STUN, period.

Regards,
 Rifaat



From gonzalo.camarillo@ericsson.com  Thu Dec 17 06:55:33 2009
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Subject: Re: [sipcore] I-D	ACTION:draft-ietf-sipcore-location-conveyance-01.txt
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Hi,

note that Keith is the shepherd of this draft. The authors tell me that 
they will submit a new revision of this draft shortly.

Cheers,

Gonzalo

Gunnar Hellstrom wrote:
>  I think it would be useful if draft-ietf-sipcore-location-conveyance moved
> on in status.
> 
> /Gunnar
> -----Original Message-----
> From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-bounces@ietf.org]
> On Behalf Of Internet-Drafts@ietf.org
> Sent: Tuesday, July 14, 2009 2:15 AM
> To: i-d-announce@ietf.org
> Cc: sipcore@ietf.org
> Subject: I-D ACTION:draft-ietf-sipcore-location-conveyance-01.txt 
> 
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> This draft is a work item of the Session Initiation Protocol Core Working
> Group of the IETF.
> 
> 	Title		: Location Conveyance for the Session Initiation
> Protocol
> 	Author(s)	: J. Polk, B. Rosen
> 	Filename	: draft-ietf-sipcore-location-conveyance-01.txt
> 	Pages		: 48
> 	Date		: 2009-7-13
> 	
> This document defines an extension to the Session Initiation 
>    Protocol (SIP) to convey geographic location information from one 
>    SIP entity to another SIP entity.  The extension covers end-to-end 
>    conveyance as well as location-based routing, where SIP servers 
>    make routing decisions based on the location of the user agent 
>    client.
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-sipcore-location-conveyance-0
> 1.txt
> 
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
> 
> Below is the data which will enable a MIME compliant mail reader
> implementation to automatically retrieve the ASCII version of the
> Internet-Draft.
> 
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 


From gonzalo.camarillo@ericsson.com  Thu Dec 17 08:08:55 2009
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Hi,

we are half-way through this WGLC and no comments have been received. 
You need to fire up! ;o)

Cheers,

Gonzalo

Gonzalo Camarillo wrote:
> Folks,
> 
> Christer has submitted a new version of the INFO Events draft (see email 
> below) that is supposed to address all the comments received so far. 
> Since there have been extensive discussions on this topic since the last 
> time we WGLCed this draft, we are going to WGLC it again.
> 
> http://tools.ietf.org/html/draft-ietf-sipcore-info-events-03
> 
> So, please, have a look at this new version and let us know whether or 
> not you also think all comments have been addressed. Of course, you are 
> also welcome to submit new comments should you have some. This WGLC will 
> end on December 24th.
> 
> Thanks,
> 
> Gonzalo
> SIPCORE co-chair
> 
> -------- Original Message --------
> Subject:     [sipcore] Draft new version: draft-ietf-sipcore-info-events
> Date:     Wed, 2 Dec 2009 09:59:24 +0100
> From:     Christer Holmberg <christer.holmberg@ericsson.com>
> To:     <sipcore@ietf.org>
> 
> 
> 
> 
> Hi,
> 
> I've submitted a new version of the info draft. Hopefully all comments
> and feedback have now been addressed.
> 
> The biggest changes are:
> 
> - Section 4.2: The usage of the Recv-Info header has been clarified,
> some simple rules have been added, and there were some discussions that
> since the re-INVITE fallback issue also affects Recv-Info (if you
> receive a Recv-Info in a 18x to a re-INVITE, and the re-INVITE then
> fails) and that we need some words about that. The text now says that IF
> the request contains a Recv-Info header, the response must also contain it.
> 
> - Section 11.4: Contains new wording about the IANA process for
> registering info packages, including the expert review etc.
> 
> - Section 12: Additional examples have been added
> 
> 
> Please note that I did NOT remove Recv-Info in PRACKs, since there in my
> opinion was no concensus to do so.
> 
> 
> The draft can also be found at:
> 
> _http://users.piuha.net/cholmber/drafts/draft-ietf-sipcore-info-events-03.txt_ 
> 
> 
> 
> 
> Regards,
> 
> Christer
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore


From vkg@alcatel-lucent.com  Tue Dec 22 09:45:09 2009
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Cullen Jennings wrote:
> On Nov 24, 2009, at 2:26 PM, Brian Hibbard wrote:
>> == Observed Interoperability Issues in section 6== Who observed
>> them? Is this experience from SIPits, etc? I think it would
>> strengthen the idea that these are real-world, observed-in-the-wild
>> issues to give sources.
> 
> Kumiko and I at SIPits are the main ref but this is all pretty uh
> fluffy, it's not like we have enough observation for any significant
> statistical sample. [...]

Here is a table that contains the number of TLS implementations
since SIPit 16:

    SIPit   Unique          Number of TLS
    No.     Implementations implementations   Percentage
    ----------------------------------------------------
    16       57                  25               44%
    18       73                  30               41%
    19       90                  41               45%
    20       90                  42               46%
    21       70                  34               49%
    22       80                  50               63%
    23       50                  24               48%
    24       43                  22               52%
    25       42                  16               38%

Note: (1) 16th SIPit was held in April 2005, 23 SIPit was held in
           October 2008.
       (2) Data for 16th SIPit is form my private archive
           and is not reflected in [1].
       (3) I don't have data for 17th SIPit.
       (4) Data from SIPit 18-25 is archived at [1].

[1] Robert J. Sparks, SIPit Summaries, archived at
     https://www.sipit.net/SIPitSummaries

In the last 3-4 bakeoffs, there weren't any implementations of
S/MIME.  In the earlier bakeoffs, there were very few implementations
of S/MIME.  If you want me to compile these for a summary, I can
do so.

I am preparing a review of sec-flows.  I will include this table
in the review comments as well.

Thanks,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
Web:   http://ect.bell-labs.com/who/vkg/

From vkg@alcatel-lucent.com  Tue Dec 22 12:28:59 2009
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First off, sorry for the delay in getting this review out.  I had
promised to send it shortly after Thanksgiving, but ...

Second, I believe that this work is of importance to the development
community that can use an additional resource to get a TLS-capable
SIP entity tested and running.

That said, here are the rest of my comments from my review.

1) S1, paragraph 2:

  OLD:
  Furthermore, testing at the events is often hampered by trying
  to get certificates signed by some common test root into the
  appropriate format for various clients.

  NEW:
  Furthermore, testing at the events is often hindered due to the
  lack of a commonly trusted certificate authority to sign the
  certificates used in the events.

2) S1, paragraph 2: The following text

    In addition, this document provides a common certificate and
    private key that can be used for a Certificate Authority (CA)
    to reduce the time it takes to set up a test at an
    interoperability event.

  makes the reader believe that the common certificate and private
  key can be used by any CA in the world.  This is, of course,
  not the intent.  The document is providing a common certificate
  and the associated private key for the express purpose of creating
  a mock CA to sign the certificates for the interoperability events.

  Thus, I would suggest the following paragraph as a substitute for
  the above one (please feel free to edit as needed):

    In addition, this document provides a common certificate and
    private key that can be used to set up a CA --- a mock CA ---
    during the SIP interoperability events.  Certificate requests from
    the users will be signed by the private key of the mock CA.

3) S1, paragraph 4, last sentence:
  s/the same CA certificate/the same mock CA private key/

4) S1, paragraph 5:
  s/list of things implementers/list of items that implementers/


5) S1, paragraph 6:
  s/A way to make/Scripts and instructions to make/
  s/is presented/are presented/

6) S3.2, note that the DNS label in the certificate under the SAN
  extension is wrong.  It should be "DNS:example.com" and not
  "DNS:com".

  Also, the example.net certificate suffers from the same problem,
  it also contains "DNS:net" in the SAN instead of "DNS:example.net".

7) S3.3, the sentence "The message could be coming from a host
  called atlanta.example.com, and the AOR in the user certificate would
  still be the same." adds ambiguity to the text without contributing
  much information.  Unless there is a specific reason why the reader
  needs to know that the SIP message sent over a TLS channel established
  by this certificate originated from the host atlanta.example.com, I
  would suggest to remove the offending sentence completely.

8) S4.1, paragraph 1:
  s/TLSRFC/TLS RFC/
  s/set to example.net/contains example.net

9) S4.1, the ssldump output: should not the second endpoint be
  identified as "www.example.net (5061)"?  See the very first line
  of the ssldump output.

10) S4.2: The first paragraph says that the <allOneLine> convention
  from rfc4475 is used, however, I do not see the <allOneLine>
  element in the subsequent example.

11) S4.2, first open issue: If Contact is not allowed in a MESSAGE
  and its corresponding 2xx, simply take it out, no?  Taking the
  Contact out does not result in any loss of generalization, nor
  does it detract from the point the example is trying to make.

12) S4.2, second open issue: If you take Contact out of the 200 OK,
  then you do not have this problem.  The Contact in the 200 OK
  contains an IP address that is the same as the one in the top-most
  Via header (which is what leads to the confusion in the first
  place, I believe.  The To and From contain URIs with domain names.)

13) S4.2, third open issue: I am not sure what you are trying to
  point out by referencing rfc3263 or connect-reuse here.  IMHO, the
  section looks fine without referencing rfc3263 or connect-reuse.

14) S5.1: I would advise being consistent and using <allOneLine>
  convention everywhere in this draft.

15) S5.1, this is a nit, but to be orthogonal to S4.2, it will help
  if the message body in S5.1 contained the same message as in
  S4.2 (i.e., "Hello!" instead of "hello").

15) S5.3: I would advise being consistent and using <allOneLine>
  convention everywhere in this draft.

16) S6: First open issue: Cullen has already stated that he and
  Kumiko has observed these.  In addition, here is a table that contains
  the number of TLS implementations since SIPit 16:

    SIPit   Unique          Number of TLS
    No.     Implementations implementations   Percentage
    ----------------------------------------------------
    16       57                  25               44%
    18       73                  30               41%
    19       90                  41               45%
    20       90                  42               46%
    21       70                  34               49%
    22       80                  50               63%
    23       50                  24               48%
    24       43                  22               52%
    25       42                  16               38%

    Note: (1) 16th SIPit was held in April 2005, 23 SIPit was held in
              October 2008.
          (2) Data for 16th SIPit is form my private archive
              and is not reflected in [1].
          (3) I don't have data for 17th SIPit.
          (4) Rest of the table created from data at [1].

    [1] Robert J. Sparks, SIPit Summaries, archived at
     https://www.sipit.net/SIPitSummaries

  In the last 3-4 bakeoffs, there weren't any implementations of
  S/MIME.  In the earlier bakeoffs, there were very few implementations
  of S/MIME.  If you want me to compile these for a summary, I can
  do so.

17) S6, second open issue: Does it matter?  What is gained by further
  categorizing the SIP clients into UAS, UAC, or a proxy?

18) S7, general comment -- some "folklore" is specified in the
  domain-certs document
  (http://tools.ietf.org/html/draft-ietf-sip-domain-certs-04), which
  is with the IESG right now.  The domain-certs document should
  be cited here as well.

18) S7, second bullet item is normatively specified in domain-certs.
  Suggest that a reference to domain-certs be included here for
  further reading.

19) S7, third open issue: I believe that IP addresses can appear in the
  SAN (rfc5280 says so.)  Their handling is specified in domain-certs (I
  believe they will appear as "DNS:192.168.0.1"; we need to have someone
  -- from pkix? -- ascertain this.  If this is the case, then their
  handling is specified in S7.1 of domain-certs.)

20) S7: For the test cases in S7, I wonder if it is worth creating test
  certificates issued by the mock CA and including these in the
  draft?  That way, implementers can simply download the certificates
  and use them for their testing.  If people think this is a good
  idea, I can spend some time doing this.

21) S7: last open issue: I think that the case of having one or more
  certificates in a path not provided or not available is a good one.
  Question is should we provide a certificate signed by a subordinate CA
  for testing to implementers or simply discuss this case?

Thanks,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
Web:   http://ect.bell-labs.com/who/vkg/

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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Richard Shockey'" <richard@shockey.us>, <sipcore@ietf.org>
References: <013401ca679d$3a926700$afb73500$@us>
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Subject: Re: [sipcore] FW: I-D	Action:draft-jones-sip-forum-fax-problem-statement-00.txt
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Folks,

I didn't see any comments regarding this draft.  What is the opinion about
publishing this as an informational RFC?

Paul

> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf Of Richard Shockey
> Sent: Tuesday, November 17, 2009 10:47 AM
> To: sipcore@ietf.org; dispatch@ietf.org
> Subject: [sipcore] FW: I-D Action:draft-jones-sip-forum-fax-problem-
> statement-00.txt
> 
> 
> FYI to the general SIP community.
> 
> There have been substantial field reports of difficulties with T.30 FAX
> is
> SIP networks and the SIP Forum was asked to help facilitate discussions
> among a group of technical experts in fax order to provider
> recommendations
> to either the IETF and or ITU SG16 on possible approaches to the
> problem.
> 
> Fax is a essential realtime communications service that is not going
> away
> and making fax work properly needs to be addressed.
> 
> -----Original Message-----
> From: i-d-announce-bounces@ietf.org [mailto:i-d-announce-
> bounces@ietf.org]
> On Behalf Of Internet-Drafts@ietf.org
> Sent: Tuesday, November 17, 2009 3:15 AM
> To: i-d-announce@ietf.org
> Subject: I-D Action:draft-jones-sip-forum-fax-problem-statement-00.txt
> 
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> 
> 	Title           : SIP Forum - Fax Over IP Task Group Problem
> Statement
> 	Author(s)       : X. Chen, et al.
> 	Filename        : draft-jones-sip-forum-fax-problem-statement-
> 00.txt
> 	Pages           : 13
> 	Date            : 2009-11-17
> 
> This memo is published for informational purposes to document the
> issues
> identified by the SIP Forum with respect to the transmission of
> facsimile
> signaling messages and fax page data over Internet Protocol (IP)
> networks.
> Further, it is the intent of this memo to alert the IETF to the
> formation of
> a Fax Over IP Task Group within the SIP Forum chartered to investigate
> and
> address identified issues as they relate to the deployment of fax
> services
> in Session Initiation Protocol (SIP) networks.
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-jones-sip-forum-fax-problem-
> statem
> ent-00.txt
> 
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
> 
> Below is the data which will enable a MIME compliant mail reader
> implementation to automatically retrieve the ASCII version of the
> Internet-Draft.
> 
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 



From dean.willis@softarmor.com  Wed Dec 23 19:19:30 2009
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Subject: Re: [sipcore] draft-ietf-sipcore-sec-flows-01: technical issues
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Vijay K. Gurbani wrote:

> Here is a table that contains the number of TLS implementations
> since SIPit 16:

The adoption rate of TLS peaked a couple of years back, and has been 
steadily declining since? That should tell us something important. Like, 
maybe nobody can figure out how to implement this stuff, nobody actually 
uses it, or nobody runs SIP on a hostile network. Not sure which is the 
case, here.

--
Dean

From dean.willis@softarmor.com  Wed Dec 23 19:24:49 2009
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Paul E. Jones wrote:
> Folks,
> 
> I didn't see any comments regarding this draft.  What is the opinion about
> publishing this as an informational RFC?

I'm sympathetic to the fax problem.

But I'm not sure what the intent of publishing this draft as an RFC 
might be. It reads more like a liaison statement from FaxTG to IETF 
informing IETF of what FaxTG is working on. We have a less costly 
pulication route for LS available that might be a better alternative if 
that's the case.

If, however, you're leading towards another fax effort in IETF, then it 
might make sense to take this draft along the individual informational 
route.

--
Dean



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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>
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Subject: Re: [sipcore] FW:	I-D	Action:draft-jones-sip-forum-fax-problem-statement-00.txt
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Dean,

The intent is not only to inform the IETF, but the public at large as to the
issues identified with fax transmission.

The next step is to try to fix the problems.  That might result in the
creation of one or more documents that prescribes a solution.  Will that be
in the ITU or IETF, or both?  I don't know.  (I'm just serving as editor.)

Let's assume the work is only done in the ITU.  Would the IETF want to
publish an Informational document at all?  It sounds like that answer might
be "no".

If the SIP Forum plans to come back to the IETF with some work, how ought we
change this document to "pave the way" for that kind of work?  I know a few
folks working in the SIP Forum are on this list, but I'll take any feedback
you and others provide to the group to ensure the message is conveyed.

Paul

PS - Merry Christmas, everyone!

> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf Of Dean Willis
> Sent: Wednesday, December 23, 2009 10:24 PM
> To: Paul E. Jones
> Cc: sipcore@ietf.org
> Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-forum-fax-
> problem-statement-00.txt
> 
> Paul E. Jones wrote:
> > Folks,
> >
> > I didn't see any comments regarding this draft.  What is the opinion
> about
> > publishing this as an informational RFC?
> 
> I'm sympathetic to the fax problem.
> 
> But I'm not sure what the intent of publishing this draft as an RFC
> might be. It reads more like a liaison statement from FaxTG to IETF
> informing IETF of what FaxTG is working on. We have a less costly
> pulication route for LS available that might be a better alternative if
> that's the case.
> 
> If, however, you're leading towards another fax effort in IETF, then it
> might make sense to take this draft along the individual informational
> route.
> 
> --
> Dean
> 
> 
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 



From pkyzivat@cisco.com  Wed Dec 23 20:04:01 2009
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Subject: Re: [sipcore] draft-ietf-sipcore-sec-flows-01: technical issues
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Dean Willis wrote:
> Vijay K. Gurbani wrote:
> 
>> Here is a table that contains the number of TLS implementations
>> since SIPit 16:
> 
> The adoption rate of TLS peaked a couple of years back, and has been 
> steadily declining since? That should tell us something important. Like, 
> maybe nobody can figure out how to implement this stuff, nobody actually 
> uses it, or nobody runs SIP on a hostile network. Not sure which is the 
> case, here.

I don't know the data - perhaps this just says something about which 
products are being brought to sipit, and *which* sipits they go to.

	Thanks,
	Paul

From shida@ntt-at.com  Wed Dec 23 23:25:31 2009
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To: Paul Kyzivat <pkyzivat@cisco.com>
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Cc: Francois Audet <audet.francois@gmail.com>, sipcore@ietf.org, R.Jesske@telekom.de
Subject: Re: [sipcore] Some comments on draft-barnes-sipcore-rfc4244bis-03
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 If the focus is about which target (AoR) is to be considered for=20
any special call handling such as voicemail (use of RFC4458?)
, so when call is forwarded the receiving end knows there is a=20
preferred identity which is to be honored, then we may have=20
some work to do with History-Info but otherwise I don't think=20
there is much more to do beside elaborating the voicemail section. =20

 Any thoughts?=20
=20
 Regards
  Shida

On Dec 7, 2009, at 2:52 AM, Paul Kyzivat wrote:

> The main tactic I know of that is currently deployed to address this =
need is to cancel the forwarded call after a few "rings", hopefully =
before it is transferred to voicemail there. But of course that doesn't =
work if you guess wrong about the number of rings, and especially if the =
call goes direct to voicemail.
>=20
> While there is no way, once I have forwarded a call to you, for me to =
*prevent* you from handling the call as voicemail, it is typically in =
the best interests of both of us. Specifically, if I am forwarding the =
call to you, but asked you *not* to send the call to your voicemail, =
then honoring that saves you the trouble handling the voicemail, and =
also keeps you in my good graces.
>=20
> So the key here is to communicate preferences for how the call should =
be handed, and define the recommended handling of calls in the presence =
of such a recommendation.
>=20
> The basics to do this have been in callerprefs for a long time. But I =
don't think they have been considered seriously in this context. I'm not =
*certain* that is sufficient to address the problem, but it seems like =
the right starting point.
>=20
> Note that this sort of approach *does not* require looking back in the =
History-Info to decide which voice mailbox to use.
>=20
> 	Thanks,
> 	Paul
>=20
>=20
> Gonzalo Camarillo wrote:
>> Hi,
>>> When you think about it, it makes sense.
>>>=20
>>> If you call me, you would expect to get my voicemail, not somebody =
else's.
>> yes, Paul's example has been discussed *many* times. So, addressing =
it in some way would be a good thing.
>> Cheers,
>> Gonzalo
>> _______________________________________________
>> 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


From gonzalo.camarillo@ericsson.com  Thu Dec 24 02:30:54 2009
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Subject: Re: [sipcore] WGLC draft-ietf-sipcore-info-events-03.txt
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Hi,

this WGLC ends today and we have not received *any* comments.

Cheers,

Gonzalo

Gonzalo Camarillo wrote:
> Hi,
> 
> we are half-way through this WGLC and no comments have been received. 
> You need to fire up! ;o)
> 
> Cheers,
> 
> Gonzalo
> 
> Gonzalo Camarillo wrote:
>> Folks,
>>
>> Christer has submitted a new version of the INFO Events draft (see email 
>> below) that is supposed to address all the comments received so far. 
>> Since there have been extensive discussions on this topic since the last 
>> time we WGLCed this draft, we are going to WGLC it again.
>>
>> http://tools.ietf.org/html/draft-ietf-sipcore-info-events-03
>>
>> So, please, have a look at this new version and let us know whether or 
>> not you also think all comments have been addressed. Of course, you are 
>> also welcome to submit new comments should you have some. This WGLC will 
>> end on December 24th.
>>
>> Thanks,
>>
>> Gonzalo
>> SIPCORE co-chair
>>
>> -------- Original Message --------
>> Subject:     [sipcore] Draft new version: draft-ietf-sipcore-info-events
>> Date:     Wed, 2 Dec 2009 09:59:24 +0100
>> From:     Christer Holmberg <christer.holmberg@ericsson.com>
>> To:     <sipcore@ietf.org>
>>
>>
>>
>>
>> Hi,
>>
>> I've submitted a new version of the info draft. Hopefully all comments
>> and feedback have now been addressed.
>>
>> The biggest changes are:
>>
>> - Section 4.2: The usage of the Recv-Info header has been clarified,
>> some simple rules have been added, and there were some discussions that
>> since the re-INVITE fallback issue also affects Recv-Info (if you
>> receive a Recv-Info in a 18x to a re-INVITE, and the re-INVITE then
>> fails) and that we need some words about that. The text now says that IF
>> the request contains a Recv-Info header, the response must also contain it.
>>
>> - Section 11.4: Contains new wording about the IANA process for
>> registering info packages, including the expert review etc.
>>
>> - Section 12: Additional examples have been added
>>
>>
>> Please note that I did NOT remove Recv-Info in PRACKs, since there in my
>> opinion was no concensus to do so.
>>
>>
>> The draft can also be found at:
>>
>> _http://users.piuha.net/cholmber/drafts/draft-ietf-sipcore-info-events-03.txt_ 
>>
>>
>>
>>
>> Regards,
>>
>> Christer
>>
>>
>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> 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
> 


From vkg@alcatel-lucent.com  Thu Dec 24 08:23:06 2009
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] draft-ietf-sipcore-sec-flows-01: technical issues
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Dean Willis wrote:
> The adoption rate of TLS peaked a couple of years back, and has been 
> steadily declining since? That should tell us something important. Like, 
> maybe nobody can figure out how to implement this stuff, nobody actually 
> uses it, or nobody runs SIP on a hostile network. Not sure which is the 
> case, here.

I don't think we can draw that general an argument that after
peaking the support for TLS is slipping now.  We probably need
more data about the early bakeoffs -- going back to the 11th
bakeoff, which is the first one that took place after rfc3261
was released.

In any case, I think the data tells us that over the 9 SIPits
during which the data has been available, the support for TLS has
hovered in [0.38 - 0.63] range, with an average of 0.47.

Essentially, less than 1/2 of the SIPit implementations that
are brought to the event support TLS.  That itself is a
telling statistic.

Thanks,

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
1960 Lucent Lane, Rm. 9C-533, Naperville, Illinois 60566 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
Web:   http://ect.bell-labs.com/who/vkg/

From richard@shockey.us  Thu Dec 24 08:43:10 2009
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From: "Richard Shockey" <richard@shockey.us>
To: "'Dean Willis'" <dean.willis@softarmor.com>, "'Paul E. Jones'" <paulej@packetizer.com>
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Cc: dispatch@ietf.org, sipcore@ietf.org
Subject: Re: [sipcore] FW:	I-D	Action:draft-jones-sip-forum-fax-problem-statement-00.txt
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The intent of the draft is to inform and clearly document to the IETF SIP
community that "there is a problem" with fax in SIP networks. In fact, the
problem seems to be much worse than some of us have originally believed. 

I recently met with several folks from the international wholesale carrier
community (the i3forum ) who are reporting similar problems with fax in
their networks. These issues have the potential for delaying significant SIP
deployment in these networks, aka loss of sales for the supplier community.
(hint)

Let us not forget that fax is a essential communications service in the PSTN
and that SIP networks MUST figure out ways to accommodate it.

The SIP Forum Fax Task group is currently investigating the problems
documented in the draft and taking additional input from all parties.
Everyone in the IETF SIP Community is free to participate in this work.

http://www.sipforum.org/content/view/310/252/

The goal is to develop a formal SIP Forum recommendation on what steps can
be taken to fix the problems. If it is determined that either SIP or T.38
protocols needs to be modified in some way then a follow on document ( ID or
ITU Liaison letter ) will be developed suggesting alternatives for
consideration.

With that follow up document in mind I do think is important to publish the
fax problem statement as Informational.


Richard Shockey
Shockey Consulting
Chairman of the Board of Directors SIP Forum
PSTN Mobile: +1 703.593.2683
<mailto:richard(at)shockey.us>
skype: rshockey101 
LinkedIn : http://www.linkedin.com/in/rshockey101
http//www.sipforum.org



>  -----Original Message-----
>  From: Dean Willis [mailto:dean.willis@softarmor.com]
>  Sent: Wednesday, December 23, 2009 10:24 PM
>  To: Paul E. Jones
>  Cc: 'Richard Shockey'; sipcore@ietf.org
>  Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-forum-fax-
>  problem-statement-00.txt
>  
>  Paul E. Jones wrote:
>  > Folks,
>  >
>  > I didn't see any comments regarding this draft.  What is the opinion
>  about
>  > publishing this as an informational RFC?
>  
>  I'm sympathetic to the fax problem.
>  
>  But I'm not sure what the intent of publishing this draft as an RFC
>  might be. It reads more like a liaison statement from FaxTG to IETF
>  informing IETF of what FaxTG is working on. We have a less costly
>  pulication route for LS available that might be a better alternative
>  if
>  that's the case.
>  
>  If, however, you're leading towards another fax effort in IETF, then
>  it
>  might make sense to take this draft along the individual informational
>  route.
>  
>  --
>  Dean



From pkyzivat@cisco.com  Thu Dec 24 12:33:58 2009
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Date: Thu, 24 Dec 2009 15:33:32 -0500
From: Paul Kyzivat <pkyzivat@cisco.com>
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Cc: "'Paul E. Jones'" <paulej@packetizer.com>, dispatch@ietf.org, sipcore@ietf.org
Subject: Re: [sipcore] [dispatch] FW:	I-D	Action:draft-jones-sip-forum-fax-problem-statement-00.txt
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I know this is a bit off topic, and doesn't alleviate the problem, but 
has anybody ever proposed that an INVITE with offer for fax be responded 
  to with a 3xx containing a mailto: URI? Or that ENUM for a fax device 
yield a mailto uri?

	Thanks,
	Paul

Richard Shockey wrote:
> The intent of the draft is to inform and clearly document to the IETF SIP
> community that "there is a problem" with fax in SIP networks. In fact, the
> problem seems to be much worse than some of us have originally believed. 
> 
> I recently met with several folks from the international wholesale carrier
> community (the i3forum ) who are reporting similar problems with fax in
> their networks. These issues have the potential for delaying significant SIP
> deployment in these networks, aka loss of sales for the supplier community.
> (hint)
> 
> Let us not forget that fax is a essential communications service in the PSTN
> and that SIP networks MUST figure out ways to accommodate it.
> 
> The SIP Forum Fax Task group is currently investigating the problems
> documented in the draft and taking additional input from all parties.
> Everyone in the IETF SIP Community is free to participate in this work.
> 
> http://www.sipforum.org/content/view/310/252/
> 
> The goal is to develop a formal SIP Forum recommendation on what steps can
> be taken to fix the problems. If it is determined that either SIP or T.38
> protocols needs to be modified in some way then a follow on document ( ID or
> ITU Liaison letter ) will be developed suggesting alternatives for
> consideration.
> 
> With that follow up document in mind I do think is important to publish the
> fax problem statement as Informational.
> 
> 
> Richard Shockey
> Shockey Consulting
> Chairman of the Board of Directors SIP Forum
> PSTN Mobile: +1 703.593.2683
> <mailto:richard(at)shockey.us>
> skype: rshockey101 
> LinkedIn : http://www.linkedin.com/in/rshockey101
> http//www.sipforum.org
> 
> 
> 
>>  -----Original Message-----
>>  From: Dean Willis [mailto:dean.willis@softarmor.com]
>>  Sent: Wednesday, December 23, 2009 10:24 PM
>>  To: Paul E. Jones
>>  Cc: 'Richard Shockey'; sipcore@ietf.org
>>  Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-forum-fax-
>>  problem-statement-00.txt
>>  
>>  Paul E. Jones wrote:
>>  > Folks,
>>  >
>>  > I didn't see any comments regarding this draft.  What is the opinion
>>  about
>>  > publishing this as an informational RFC?
>>  
>>  I'm sympathetic to the fax problem.
>>  
>>  But I'm not sure what the intent of publishing this draft as an RFC
>>  might be. It reads more like a liaison statement from FaxTG to IETF
>>  informing IETF of what FaxTG is working on. We have a less costly
>>  pulication route for LS available that might be a better alternative
>>  if
>>  that's the case.
>>  
>>  If, however, you're leading towards another fax effort in IETF, then
>>  it
>>  might make sense to take this draft along the individual informational
>>  route.
>>  
>>  --
>>  Dean
> 
> 
> _______________________________________________
> dispatch mailing list
> dispatch@ietf.org
> https://www.ietf.org/mailman/listinfo/dispatch
> 

From paulej@packetizer.com  Thu Dec 24 13:27:34 2009
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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Paul Kyzivat'" <pkyzivat@cisco.com>, "'Richard Shockey'" <richard@shockey.us>
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Cc: dispatch@ietf.org, sipcore@ietf.org
Subject: Re: [sipcore] [dispatch] I-D Action:draft-jones-sip-forum-fax-problem-statement-00.txt
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Paul,

That could certainly be done and T.37 could be used on gateways to make that
work.

But, as you said, it does not fix the problem with T.38.  Perhaps equally
important, I think there are many, many businesses that will not be moving
to email anytime soon.  I wish they would -- that would solve this problem
:-)
 
Paul

> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf Of Paul Kyzivat
> Sent: Thursday, December 24, 2009 3:34 PM
> To: Richard Shockey
> Cc: 'Paul E. Jones'; dispatch@ietf.org; sipcore@ietf.org
> Subject: Re: [sipcore] [dispatch] FW: I-D Action:draft-jones-sip-forum-
> fax-problem-statement-00.txt
> 
> I know this is a bit off topic, and doesn't alleviate the problem, but
> has anybody ever proposed that an INVITE with offer for fax be
> responded
>   to with a 3xx containing a mailto: URI? Or that ENUM for a fax device
> yield a mailto uri?
> 
> 	Thanks,
> 	Paul
> 
> Richard Shockey wrote:
> > The intent of the draft is to inform and clearly document to the IETF
> SIP
> > community that "there is a problem" with fax in SIP networks. In
> fact, the
> > problem seems to be much worse than some of us have originally
> believed.
> >
> > I recently met with several folks from the international wholesale
> carrier
> > community (the i3forum ) who are reporting similar problems with fax
> in
> > their networks. These issues have the potential for delaying
> significant SIP
> > deployment in these networks, aka loss of sales for the supplier
> community.
> > (hint)
> >
> > Let us not forget that fax is a essential communications service in
> the PSTN
> > and that SIP networks MUST figure out ways to accommodate it.
> >
> > The SIP Forum Fax Task group is currently investigating the problems
> > documented in the draft and taking additional input from all parties.
> > Everyone in the IETF SIP Community is free to participate in this
> work.
> >
> > http://www.sipforum.org/content/view/310/252/
> >
> > The goal is to develop a formal SIP Forum recommendation on what
> steps can
> > be taken to fix the problems. If it is determined that either SIP or
> T.38
> > protocols needs to be modified in some way then a follow on document
> ( ID or
> > ITU Liaison letter ) will be developed suggesting alternatives for
> > consideration.
> >
> > With that follow up document in mind I do think is important to
> publish the
> > fax problem statement as Informational.
> >
> >
> > Richard Shockey
> > Shockey Consulting
> > Chairman of the Board of Directors SIP Forum
> > PSTN Mobile: +1 703.593.2683
> > <mailto:richard(at)shockey.us>
> > skype: rshockey101
> > LinkedIn : http://www.linkedin.com/in/rshockey101
> > http//www.sipforum.org
> >
> >
> >
> >>  -----Original Message-----
> >>  From: Dean Willis [mailto:dean.willis@softarmor.com]
> >>  Sent: Wednesday, December 23, 2009 10:24 PM
> >>  To: Paul E. Jones
> >>  Cc: 'Richard Shockey'; sipcore@ietf.org
> >>  Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-forum-fax-
> >>  problem-statement-00.txt
> >>
> >>  Paul E. Jones wrote:
> >>  > Folks,
> >>  >
> >>  > I didn't see any comments regarding this draft.  What is the
> opinion
> >>  about
> >>  > publishing this as an informational RFC?
> >>
> >>  I'm sympathetic to the fax problem.
> >>
> >>  But I'm not sure what the intent of publishing this draft as an RFC
> >>  might be. It reads more like a liaison statement from FaxTG to IETF
> >>  informing IETF of what FaxTG is working on. We have a less costly
> >>  pulication route for LS available that might be a better
> alternative
> >>  if
> >>  that's the case.
> >>
> >>  If, however, you're leading towards another fax effort in IETF,
> then
> >>  it
> >>  might make sense to take this draft along the individual
> informational
> >>  route.
> >>
> >>  --
> >>  Dean
> >
> >
> > _______________________________________________
> > dispatch mailing list
> > dispatch@ietf.org
> > https://www.ietf.org/mailman/listinfo/dispatch
> >
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 



From eburger@standardstrack.com  Fri Dec 25 04:10:20 2009
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] WGLC draft-ietf-sipcore-info-events-03.txt
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I guess that means that after two years we're perfect? :-)

On Dec 24, 2009, at 5:30 AM, Gonzalo Camarillo wrote:

> Hi,
>
> this WGLC ends today and we have not received *any* comments.
>
> Cheers,
>
> Gonzalo
>
> Gonzalo Camarillo wrote:
>> Hi,
>> we are half-way through this WGLC and no comments have been  
>> received. You need to fire up! ;o)
>> Cheers,
>> Gonzalo
>> Gonzalo Camarillo wrote:
>>> Folks,
>>>
>>> Christer has submitted a new version of the INFO Events draft (see  
>>> email below) that is supposed to address all the comments received  
>>> so far. Since there have been extensive discussions on this topic  
>>> since the last time we WGLCed this draft, we are going to WGLC it  
>>> again.
>>>
>>> http://tools.ietf.org/html/draft-ietf-sipcore-info-events-03
>>>
>>> So, please, have a look at this new version and let us know  
>>> whether or not you also think all comments have been addressed. Of  
>>> course, you are also welcome to submit new comments should you  
>>> have some. This WGLC will end on December 24th.
>>>
>>> Thanks,
>>>
>>> Gonzalo
>>> SIPCORE co-chair
>>>
>>> -------- Original Message --------
>>> Subject:     [sipcore] Draft new version: draft-ietf-sipcore-info- 
>>> events
>>> Date:     Wed, 2 Dec 2009 09:59:24 +0100
>>> From:     Christer Holmberg <christer.holmberg@ericsson.com>
>>> To:     <sipcore@ietf.org>
>>>
>>>
>>>
>>>
>>> Hi,
>>>
>>> I've submitted a new version of the info draft. Hopefully all  
>>> comments
>>> and feedback have now been addressed.
>>>
>>> The biggest changes are:
>>>
>>> - Section 4.2: The usage of the Recv-Info header has been clarified,
>>> some simple rules have been added, and there were some discussions  
>>> that
>>> since the re-INVITE fallback issue also affects Recv-Info (if you
>>> receive a Recv-Info in a 18x to a re-INVITE, and the re-INVITE then
>>> fails) and that we need some words about that. The text now says  
>>> that IF
>>> the request contains a Recv-Info header, the response must also  
>>> contain it.
>>>
>>> - Section 11.4: Contains new wording about the IANA process for
>>> registering info packages, including the expert review etc.
>>>
>>> - Section 12: Additional examples have been added
>>>
>>>
>>> Please note that I did NOT remove Recv-Info in PRACKs, since there  
>>> in my
>>> opinion was no concensus to do so.
>>>
>>>
>>> The draft can also be found at:
>>>
>>> _http://users.piuha.net/cholmber/drafts/draft-ietf-sipcore-info-events-03.txt_
>>>
>>>
>>>
>>> Regards,
>>>
>>> Christer
>>>
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> 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


From richard@shockey.us  Sat Dec 26 09:29:46 2009
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From: "Richard Shockey" <richard@shockey.us>
To: "'Paul Kyzivat'" <pkyzivat@cisco.com>
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Cc: "'Paul E. Jones'" <paulej@packetizer.com>, dispatch@ietf.org, sipcore@ietf.org
Subject: Re: [sipcore] [dispatch] FW:	I-D	Action:draft-jones-sip-forum-fax-problem-statement-00.txt
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Paul its certainly not off topic and a reasonable idea, but not practical
for a number of reasons. What you describe is a classic T.37 mode for fax
that the old IETF Fax WG documented in RFC 3965. The problem is "non
repudiation". Fax has a certain legal standing and there is the requirement
that there be actual documentation that a fax was sent by foo at such and
such a time to bar and that the service provider or appropriate intermediary
can provide the appropriate documentation of such a transaction.

The problem is becoming more acute due to a variety of regulations, HIPPA
and USA SOX, real estate etal which require the transmission of a signature
as part of a legal filing and no, digital signatures have not proven to be
successful even though they have legal standing in most countries now.
Health care records cannot be sent by email for privacy reasons especially
test results.

Fax is pretty simple and it works for these class of transactions. As a
result my contacts in the fax industry have noted a minor resurgence in fax
machine and fax port sales. 

Some of us years ago thought using a variant of the Internet Print Protocol,
based on HTTP for real time document transmission would have been a better
idea but that went nowhere as well.

The issue is there something in SIP signaling that can alert the first hop
what this session really is such as ;USER=FAX that can help. I don't know,
that is what needs to be investigated.

>  -----Original Message-----
>  From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>  Sent: Thursday, December 24, 2009 3:34 PM
>  To: Richard Shockey
>  Cc: 'Dean Willis'; 'Paul E. Jones'; dispatch@ietf.org;
>  sipcore@ietf.org
>  Subject: Re: [dispatch] [sipcore] FW: I-D Action:draft-jones-sip-
>  forum-fax-problem-statement-00.txt
>  
>  I know this is a bit off topic, and doesn't alleviate the problem, but
>  has anybody ever proposed that an INVITE with offer for fax be
>  responded
>    to with a 3xx containing a mailto: URI? Or that ENUM for a fax
>  device
>  yield a mailto uri?
>  
>  	Thanks,
>  	Paul
>  
>  Richard Shockey wrote:
>  > The intent of the draft is to inform and clearly document to the
>  IETF SIP
>  > community that "there is a problem" with fax in SIP networks. In
>  fact, the
>  > problem seems to be much worse than some of us have originally
>  believed.
>  >
>  > I recently met with several folks from the international wholesale
>  carrier
>  > community (the i3forum ) who are reporting similar problems with fax
>  in
>  > their networks. These issues have the potential for delaying
>  significant SIP
>  > deployment in these networks, aka loss of sales for the supplier
>  community.
>  > (hint)
>  >
>  > Let us not forget that fax is a essential communications service in
>  the PSTN
>  > and that SIP networks MUST figure out ways to accommodate it.
>  >
>  > The SIP Forum Fax Task group is currently investigating the problems
>  > documented in the draft and taking additional input from all
>  parties.
>  > Everyone in the IETF SIP Community is free to participate in this
>  work.
>  >
>  > http://www.sipforum.org/content/view/310/252/
>  >
>  > The goal is to develop a formal SIP Forum recommendation on what
>  steps can
>  > be taken to fix the problems. If it is determined that either SIP or
>  T.38
>  > protocols needs to be modified in some way then a follow on document
>  ( ID or
>  > ITU Liaison letter ) will be developed suggesting alternatives for
>  > consideration.
>  >
>  > With that follow up document in mind I do think is important to
>  publish the
>  > fax problem statement as Informational.
>  >
>  >
>  > Richard Shockey
>  > Shockey Consulting
>  > Chairman of the Board of Directors SIP Forum
>  > PSTN Mobile: +1 703.593.2683
>  > <mailto:richard(at)shockey.us>
>  > skype: rshockey101
>  > LinkedIn : http://www.linkedin.com/in/rshockey101
>  > http//www.sipforum.org
>  >
>  >
>  >
>  >>  -----Original Message-----
>  >>  From: Dean Willis [mailto:dean.willis@softarmor.com]
>  >>  Sent: Wednesday, December 23, 2009 10:24 PM
>  >>  To: Paul E. Jones
>  >>  Cc: 'Richard Shockey'; sipcore@ietf.org
>  >>  Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-forum-fax-
>  >>  problem-statement-00.txt
>  >>
>  >>  Paul E. Jones wrote:
>  >>  > Folks,
>  >>  >
>  >>  > I didn't see any comments regarding this draft.  What is the
>  opinion
>  >>  about
>  >>  > publishing this as an informational RFC?
>  >>
>  >>  I'm sympathetic to the fax problem.
>  >>
>  >>  But I'm not sure what the intent of publishing this draft as an
>  RFC
>  >>  might be. It reads more like a liaison statement from FaxTG to
>  IETF
>  >>  informing IETF of what FaxTG is working on. We have a less costly
>  >>  pulication route for LS available that might be a better
>  alternative
>  >>  if
>  >>  that's the case.
>  >>
>  >>  If, however, you're leading towards another fax effort in IETF,
>  then
>  >>  it
>  >>  might make sense to take this draft along the individual
>  informational
>  >>  route.
>  >>
>  >>  --
>  >>  Dean
>  >
>  >
>  > _______________________________________________
>  > dispatch mailing list
>  > dispatch@ietf.org
>  > https://www.ietf.org/mailman/listinfo/dispatch
>  >


From paulej@packetizer.com  Sat Dec 26 12:48:04 2009
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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Richard Shockey'" <richard@shockey.us>, "'Paul Kyzivat'" <pkyzivat@cisco.com>
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Cc: sipcore@ietf.org
Subject: Re: [sipcore] [dispatch] I-D Action:draft-jones-sip-forum-fax-problem-statement-00.txt
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Richard,

All of your points are correct, though I think the biggest reason why fax is
important is that fax is simple and works.  (Well, except that it does not
work well on SIP networks, thus the reason we're discussing this.)

The statement about legal standing of faxes is something I've often heard,
but I've been curious about it.  How could a fax document have any legal
standing when I can so easily (and legally) forge the phone number I use to
place the call?  If "legal reasons" are the primary reason for using fax, I
believe we could find a more modern solution to the need to transmit paper
documents.

I think it's important to solve the fax issues, because that's the
technology people have in their hands right now.  That said, I'll also tell
you that I really, really like those new "scan / PDF / email" machines we
have in our office.  I can't say enough good words about those machines and
I now use those (or just a simple scanner) when I need to sign and transmit
legal documents.  I've not used a fax machine in years.  While there is a
non-repudiation issue, that bothers me very little since I know someone
could easily take my signature and insert it into a document and fax it.  At
the end of the day, issues can only be resolved by having a person swear
under oath that he/she did or did not produce/sign/transmit some document.

I think our society sometimes goes overboard with their perception of
"security".  People do not fully understand that most security measures are
not "secure" but just assurance levels.  As evidence to this misguided
thought, I went to the bank one day this year to withdraw money.  The teller
did not like my signature because it did not match the one she had on file
from 10 years ago. "Forget about the signature: it's not that important", I
told her.  She could not believe I would utter such a thing.  As far as she
was concerned, that signature meant everything in the world in terms of
security.  For me, though, my identification and me standing right before
you has more weight.  The lady who works by the front door of the bank has
been there since I started banking there and she knows me by name, so I told
the teller to ask her to come over and verify my identity.  She refused.
She put more trust in that signature than she would in my face, my
identification card, and the word of an employee who knew me as decade-long
customer.  But, I bet she was just following some misguided directions in
the name of "security".

The point is that the tools we put into place only help to provide some
level of assurances, and fax is relatively weak in that regard.  It's even
worse than my signature, since so many aspects of that document can be
forged.  (Heck, a pair of alligator clips are the only tools one needs to
intercept PSTN calls, and it takes little more than that to alter the
transmitted fax data.)

So, I that we really have two issues:
1) We need to resolve the fax transmission issues in SIP networks, because
fax is being used by many, many businesses and might even be the primary
business tool for some businesses.  How long will this be the norm?
2) We need to explore alternatives that can provide whatever reasonable
assurances we can.  It seems the market is setting the direction here
(emailed PDF documents), as I am seeing this capability in relatively
low-end machines at Office Depot now.  Perhaps this is sufficient, as we now
have tools like DKIM, S/MIME, etc. to assist with the assurance aspects.

What Paul K. was suggesting, I guess, was to start migrating from (1) to (2)
through the use of NAPTR records.  I think it's a reasonable thing to
consider, as I believe the world is moving to (2), anyway.  I just doubt the
world will move entirely to email or any other transmission system for a few
years, so I don't want us to be distracted from the immediate problem at
hand, else we'll be fighting this another 10 years.

Paul

> -----Original Message-----
> From: Richard Shockey [mailto:richard@shockey.us]
> Sent: Saturday, December 26, 2009 12:23 PM
> To: 'Paul Kyzivat'
> Cc: 'Dean Willis'; 'Paul E. Jones'; dispatch@ietf.org; sipcore@ietf.org
> Subject: RE: [dispatch] [sipcore] FW: I-D Action:draft-jones-sip-forum-
> fax-problem-statement-00.txt
> 
> Paul its certainly not off topic and a reasonable idea, but not
> practical
> for a number of reasons. What you describe is a classic T.37 mode for
> fax
> that the old IETF Fax WG documented in RFC 3965. The problem is "non
> repudiation". Fax has a certain legal standing and there is the
> requirement
> that there be actual documentation that a fax was sent by foo at such
> and
> such a time to bar and that the service provider or appropriate
> intermediary
> can provide the appropriate documentation of such a transaction.
> 
> The problem is becoming more acute due to a variety of regulations,
> HIPPA
> and USA SOX, real estate etal which require the transmission of a
> signature
> as part of a legal filing and no, digital signatures have not proven to
> be
> successful even though they have legal standing in most countries now.
> Health care records cannot be sent by email for privacy reasons
> especially
> test results.
> 
> Fax is pretty simple and it works for these class of transactions. As a
> result my contacts in the fax industry have noted a minor resurgence in
> fax
> machine and fax port sales.
> 
> Some of us years ago thought using a variant of the Internet Print
> Protocol,
> based on HTTP for real time document transmission would have been a
> better
> idea but that went nowhere as well.
> 
> The issue is there something in SIP signaling that can alert the first
> hop
> what this session really is such as ;USER=FAX that can help. I don't
> know,
> that is what needs to be investigated.
> 
> >  -----Original Message-----
> >  From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> >  Sent: Thursday, December 24, 2009 3:34 PM
> >  To: Richard Shockey
> >  Cc: 'Dean Willis'; 'Paul E. Jones'; dispatch@ietf.org;
> >  sipcore@ietf.org
> >  Subject: Re: [dispatch] [sipcore] FW: I-D Action:draft-jones-sip-
> >  forum-fax-problem-statement-00.txt
> >
> >  I know this is a bit off topic, and doesn't alleviate the problem,
> but
> >  has anybody ever proposed that an INVITE with offer for fax be
> >  responded
> >    to with a 3xx containing a mailto: URI? Or that ENUM for a fax
> >  device
> >  yield a mailto uri?
> >
> >  	Thanks,
> >  	Paul
> >
> >  Richard Shockey wrote:
> >  > The intent of the draft is to inform and clearly document to the
> >  IETF SIP
> >  > community that "there is a problem" with fax in SIP networks. In
> >  fact, the
> >  > problem seems to be much worse than some of us have originally
> >  believed.
> >  >
> >  > I recently met with several folks from the international wholesale
> >  carrier
> >  > community (the i3forum ) who are reporting similar problems with
> fax
> >  in
> >  > their networks. These issues have the potential for delaying
> >  significant SIP
> >  > deployment in these networks, aka loss of sales for the supplier
> >  community.
> >  > (hint)
> >  >
> >  > Let us not forget that fax is a essential communications service
> in
> >  the PSTN
> >  > and that SIP networks MUST figure out ways to accommodate it.
> >  >
> >  > The SIP Forum Fax Task group is currently investigating the
> problems
> >  > documented in the draft and taking additional input from all
> >  parties.
> >  > Everyone in the IETF SIP Community is free to participate in this
> >  work.
> >  >
> >  > http://www.sipforum.org/content/view/310/252/
> >  >
> >  > The goal is to develop a formal SIP Forum recommendation on what
> >  steps can
> >  > be taken to fix the problems. If it is determined that either SIP
> or
> >  T.38
> >  > protocols needs to be modified in some way then a follow on
> document
> >  ( ID or
> >  > ITU Liaison letter ) will be developed suggesting alternatives for
> >  > consideration.
> >  >
> >  > With that follow up document in mind I do think is important to
> >  publish the
> >  > fax problem statement as Informational.
> >  >
> >  >
> >  > Richard Shockey
> >  > Shockey Consulting
> >  > Chairman of the Board of Directors SIP Forum
> >  > PSTN Mobile: +1 703.593.2683
> >  > <mailto:richard(at)shockey.us>
> >  > skype: rshockey101
> >  > LinkedIn : http://www.linkedin.com/in/rshockey101
> >  > http//www.sipforum.org
> >  >
> >  >
> >  >
> >  >>  -----Original Message-----
> >  >>  From: Dean Willis [mailto:dean.willis@softarmor.com]
> >  >>  Sent: Wednesday, December 23, 2009 10:24 PM
> >  >>  To: Paul E. Jones
> >  >>  Cc: 'Richard Shockey'; sipcore@ietf.org
> >  >>  Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-forum-fax-
> >  >>  problem-statement-00.txt
> >  >>
> >  >>  Paul E. Jones wrote:
> >  >>  > Folks,
> >  >>  >
> >  >>  > I didn't see any comments regarding this draft.  What is the
> >  opinion
> >  >>  about
> >  >>  > publishing this as an informational RFC?
> >  >>
> >  >>  I'm sympathetic to the fax problem.
> >  >>
> >  >>  But I'm not sure what the intent of publishing this draft as an
> >  RFC
> >  >>  might be. It reads more like a liaison statement from FaxTG to
> >  IETF
> >  >>  informing IETF of what FaxTG is working on. We have a less
> costly
> >  >>  pulication route for LS available that might be a better
> >  alternative
> >  >>  if
> >  >>  that's the case.
> >  >>
> >  >>  If, however, you're leading towards another fax effort in IETF,
> >  then
> >  >>  it
> >  >>  might make sense to take this draft along the individual
> >  informational
> >  >>  route.
> >  >>
> >  >>  --
> >  >>  Dean
> >  >
> >  >
> >  > _______________________________________________
> >  > dispatch mailing list
> >  > dispatch@ietf.org
> >  > https://www.ietf.org/mailman/listinfo/dispatch
> >  >
> 



From richard@shockey.us  Sun Dec 27 12:58:11 2009
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From: "Richard Shockey" <richard@shockey.us>
To: "'Paul E. Jones'" <paulej@packetizer.com>, "'Paul Kyzivat'" <pkyzivat@cisco.com>
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Cc: sipcore@ietf.org
Subject: Re: [sipcore] [dispatch] I-D Action:draft-jones-sip-forum-fax-problem-statement-00.txt
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>  Richard,
>  
>  All of your points are correct, though I think the biggest reason why
>  fax is  important is that fax is simple and works.  (Well, except that it
does
>  not  work well on SIP networks, thus the reason we're discussing this.)

Exactly .. its an essential PSTN communications service that SIP networks
must be able to support. Its clean simple and its works ..just like phone
numbers.

 
>  The statement about legal standing of faxes is something I've often
>  heard,  but I've been curious about it.  How could a fax document have
any
>  legal  standing when I can so easily (and legally) forge the phone number
I
>  use to  place the call?  If "legal reasons" are the primary reason for
using
>  fax, I  believe we could find a more modern solution to the need to
transmit
>  paper documents.

I know .. but it's the 3rd party validation of the transaction through
carrier networks that carries some of the legal weight. Though with some
checking I found out that in the US because of the Uniform Electronic Clear
Transactions Act of 1999, and various legislature passed since then,
contracts are no longer limited to pristine original copies on clean, white
paper. Fax is preferred to email for health care transactions due the
perceived notion that fax is more "secure" a point to point communication
without intermediaries.

In fact, according to the UETA, email and fax contracts are now legally
binding. This means that any agreement you make through any medium -
electronic or otherwise - will hold up in court. As for alternative real
time transport I had this argument 15 years ago in the FAX WG and as I
mentioned HTTP would be much more useful transport for documents but alas
for reasons I don't need to go into that did not happen.

>  
>  I think it's important to solve the fax issues, because that's the
>  technology people have in their hands right now.  That said, I'll also
>  tell  you that I really, really like those new "scan / PDF / email"
machines
>  we  have in our office.  I can't say enough good words about those
>  machines and  I now use those (or just a simple scanner) when I need to
sign and
>  transmit legal documents.  

Yep I worked on one of the originals as a consultant years ago ..HP's
Digital Sender. Great device still have one in my office gathering dust.


>  I think our society sometimes goes overboard with their perception of
>  "security".  People do not fully understand that most security
>  measures are  not "secure" but just assurance levels.  As evidence to
this misguided
>  thought, I went to the bank one day this year to withdraw money.  The
>  teller  did not like my signature because it did not match the one she
had on
>  file  from 10 years ago. "Forget about the signature: it's not that
>  important", I>  told her.  She could not believe I would utter such a
thing.  As far
>  as she  was concerned, that signature meant everything in the world in
terms
>  of  security.  

Yep that the key ..

>  
>  So, I that we really have two issues:
>  1) We need to resolve the fax transmission issues in SIP networks,
>  because  fax is being used by many, many businesses and might even be the
>  primary  business tool for some businesses.  How long will this be the
norm?

Probably till we're dead. 


>  2) We need to explore alternatives that can provide whatever
>  reasonable>  assurances we can.  It seems the market is setting the
direction here
>  (emailed PDF documents), as I am seeing this capability in relatively
>  low-end machines at Office Depot now.  Perhaps this is sufficient, as
>  we now  have tools like DKIM, S/MIME, etc. to assist with the assurance
>  aspects.


We're geeks ..we think that is the direction the market is going. The
reality of carrier networks is very different and don't get me started on
DKIM AND S/MIME as solutions. 

>  
>  What Paul K. was suggesting, I guess, was to start migrating from (1)
>  to (2)  through the use of NAPTR records. 

You want a lecture on why RFC 3761 [ENUM] in e164.arpa failed? I regularly
give lectures on that subject. To Paul's point it would be trivial but even
private carrier enum systems have their deployment issues.

 I think it's a reasonable thing to
>  consider, as I believe the world is moving to (2), anyway.  I just
>  doubt the  world will move entirely to email or any other transmission
system for
>  a few years, so I don't want us to be distracted from the immediate
problem
>  at hand, else we'll be fighting this another 10 years.


To your point the problem at hand is that SIP carrier networks still need to
handle T.30 fax and ultimately they are paying a lot of our salaries. I
thing the FAX task group in the SIP Forum has made a excellent start but
they still need input.

>  
>  Paul
>  
>  > -----Original Message-----
>  > From: Richard Shockey [mailto:richard@shockey.us]
>  > Sent: Saturday, December 26, 2009 12:23 PM
>  > To: 'Paul Kyzivat'
>  > Cc: 'Dean Willis'; 'Paul E. Jones'; dispatch@ietf.org;
>  sipcore@ietf.org
>  > Subject: RE: [dispatch] [sipcore] FW: I-D Action:draft-jones-sip-
>  forum-
>  > fax-problem-statement-00.txt
>  >
>  > Paul its certainly not off topic and a reasonable idea, but not
>  > practical
>  > for a number of reasons. What you describe is a classic T.37 mode
>  for
>  > fax
>  > that the old IETF Fax WG documented in RFC 3965. The problem is "non
>  > repudiation". Fax has a certain legal standing and there is the
>  > requirement
>  > that there be actual documentation that a fax was sent by foo at
>  such
>  > and
>  > such a time to bar and that the service provider or appropriate
>  > intermediary
>  > can provide the appropriate documentation of such a transaction.
>  >
>  > The problem is becoming more acute due to a variety of regulations,
>  > HIPPA
>  > and USA SOX, real estate etal which require the transmission of a
>  > signature
>  > as part of a legal filing and no, digital signatures have not proven
>  to
>  > be
>  > successful even though they have legal standing in most countries
>  now.
>  > Health care records cannot be sent by email for privacy reasons
>  > especially
>  > test results.
>  >
>  > Fax is pretty simple and it works for these class of transactions.
>  As a
>  > result my contacts in the fax industry have noted a minor resurgence
>  in
>  > fax
>  > machine and fax port sales.
>  >
>  > Some of us years ago thought using a variant of the Internet Print
>  > Protocol,
>  > based on HTTP for real time document transmission would have been a
>  > better
>  > idea but that went nowhere as well.
>  >
>  > The issue is there something in SIP signaling that can alert the
>  first
>  > hop
>  > what this session really is such as ;USER=FAX that can help. I don't
>  > know,
>  > that is what needs to be investigated.
>  >
>  > >  -----Original Message-----
>  > >  From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>  > >  Sent: Thursday, December 24, 2009 3:34 PM
>  > >  To: Richard Shockey
>  > >  Cc: 'Dean Willis'; 'Paul E. Jones'; dispatch@ietf.org;
>  > >  sipcore@ietf.org
>  > >  Subject: Re: [dispatch] [sipcore] FW: I-D Action:draft-jones-sip-
>  > >  forum-fax-problem-statement-00.txt
>  > >
>  > >  I know this is a bit off topic, and doesn't alleviate the
>  problem,
>  > but
>  > >  has anybody ever proposed that an INVITE with offer for fax be
>  > >  responded
>  > >    to with a 3xx containing a mailto: URI? Or that ENUM for a fax
>  > >  device
>  > >  yield a mailto uri?
>  > >
>  > >  	Thanks,
>  > >  	Paul
>  > >
>  > >  >
>  > >  > Richard Shockey
>  > >  > Shockey Consulting
>  > >  > Chairman of the Board of Directors SIP Forum
>  > >  > PSTN Mobile: +1 703.593.2683
>  > >  > <mailto:richard(at)shockey.us>
>  > >  > skype: rshockey101
>  > >  > LinkedIn : http://www.linkedin.com/in/rshockey101
>  > >  > http//www.sipforum.org



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From: Paul Kyzivat <pkyzivat@cisco.com>
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Cc: "'Paul E. Jones'" <paulej@packetizer.com>, sipcore@ietf.org
Subject: Re: [sipcore] [dispatch] I-D Action:draft-jones-sip-forum-fax-problem-statement-00.txt
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I'm not going to pretend that I know the issues here.
Some of them seem to be irrational, but that doesn't make them any less 
real. (Surely nobody considers government and law making rational.)

What I had in mind was using email to transmit the same bits that would 
otherwise be transmitted via T.38. That does involve *different* 
intermediaries, but probably not *more* of them than transmitting via 
sip. It would be another optimization above that to transmit the doc in 
a more symbolic encoding.

I suppose the real issue is *which* intermediaries, and who is in a 
position to vouch for their security. It they are all controlled by same 
service providers it wouldn't really matter, would it?

Suppose we were considering the case of traditional fax machines on both 
ends, connected via traditional telephony infrastructure to gateways 
that then communicate via sip. In such a case, would it really matter if 
the gateways decided to transmit fax using email rather than sip?

	Thanks,
	Paul

Richard Shockey wrote:
>>  Richard,
>>  
>>  All of your points are correct, though I think the biggest reason why
>>  fax is  important is that fax is simple and works.  (Well, except that it
> does
>>  not  work well on SIP networks, thus the reason we're discussing this.)
> 
> Exactly .. its an essential PSTN communications service that SIP networks
> must be able to support. Its clean simple and its works ..just like phone
> numbers.
> 
>  
>>  The statement about legal standing of faxes is something I've often
>>  heard,  but I've been curious about it.  How could a fax document have
> any
>>  legal  standing when I can so easily (and legally) forge the phone number
> I
>>  use to  place the call?  If "legal reasons" are the primary reason for
> using
>>  fax, I  believe we could find a more modern solution to the need to
> transmit
>>  paper documents.
> 
> I know .. but it's the 3rd party validation of the transaction through
> carrier networks that carries some of the legal weight. Though with some
> checking I found out that in the US because of the Uniform Electronic Clear
> Transactions Act of 1999, and various legislature passed since then,
> contracts are no longer limited to pristine original copies on clean, white
> paper. Fax is preferred to email for health care transactions due the
> perceived notion that fax is more "secure" a point to point communication
> without intermediaries.
> 
> In fact, according to the UETA, email and fax contracts are now legally
> binding. This means that any agreement you make through any medium -
> electronic or otherwise - will hold up in court. As for alternative real
> time transport I had this argument 15 years ago in the FAX WG and as I
> mentioned HTTP would be much more useful transport for documents but alas
> for reasons I don't need to go into that did not happen.
> 
>>  
>>  I think it's important to solve the fax issues, because that's the
>>  technology people have in their hands right now.  That said, I'll also
>>  tell  you that I really, really like those new "scan / PDF / email"
> machines
>>  we  have in our office.  I can't say enough good words about those
>>  machines and  I now use those (or just a simple scanner) when I need to
> sign and
>>  transmit legal documents.  
> 
> Yep I worked on one of the originals as a consultant years ago ..HP's
> Digital Sender. Great device still have one in my office gathering dust.
> 
> 
>>  I think our society sometimes goes overboard with their perception of
>>  "security".  People do not fully understand that most security
>>  measures are  not "secure" but just assurance levels.  As evidence to
> this misguided
>>  thought, I went to the bank one day this year to withdraw money.  The
>>  teller  did not like my signature because it did not match the one she
> had on
>>  file  from 10 years ago. "Forget about the signature: it's not that
>>  important", I>  told her.  She could not believe I would utter such a
> thing.  As far
>>  as she  was concerned, that signature meant everything in the world in
> terms
>>  of  security.  
> 
> Yep that the key ..
> 
>>  
>>  So, I that we really have two issues:
>>  1) We need to resolve the fax transmission issues in SIP networks,
>>  because  fax is being used by many, many businesses and might even be the
>>  primary  business tool for some businesses.  How long will this be the
> norm?
> 
> Probably till we're dead. 
> 
> 
>>  2) We need to explore alternatives that can provide whatever
>>  reasonable>  assurances we can.  It seems the market is setting the
> direction here
>>  (emailed PDF documents), as I am seeing this capability in relatively
>>  low-end machines at Office Depot now.  Perhaps this is sufficient, as
>>  we now  have tools like DKIM, S/MIME, etc. to assist with the assurance
>>  aspects.
> 
> 
> We're geeks ..we think that is the direction the market is going. The
> reality of carrier networks is very different and don't get me started on
> DKIM AND S/MIME as solutions. 
> 
>>  
>>  What Paul K. was suggesting, I guess, was to start migrating from (1)
>>  to (2)  through the use of NAPTR records. 
> 
> You want a lecture on why RFC 3761 [ENUM] in e164.arpa failed? I regularly
> give lectures on that subject. To Paul's point it would be trivial but even
> private carrier enum systems have their deployment issues.
> 
>  I think it's a reasonable thing to
>>  consider, as I believe the world is moving to (2), anyway.  I just
>>  doubt the  world will move entirely to email or any other transmission
> system for
>>  a few years, so I don't want us to be distracted from the immediate
> problem
>>  at hand, else we'll be fighting this another 10 years.
> 
> 
> To your point the problem at hand is that SIP carrier networks still need to
> handle T.30 fax and ultimately they are paying a lot of our salaries. I
> thing the FAX task group in the SIP Forum has made a excellent start but
> they still need input.
> 
>>  
>>  Paul
>>  
>>  > -----Original Message-----
>>  > From: Richard Shockey [mailto:richard@shockey.us]
>>  > Sent: Saturday, December 26, 2009 12:23 PM
>>  > To: 'Paul Kyzivat'
>>  > Cc: 'Dean Willis'; 'Paul E. Jones'; dispatch@ietf.org;
>>  sipcore@ietf.org
>>  > Subject: RE: [dispatch] [sipcore] FW: I-D Action:draft-jones-sip-
>>  forum-
>>  > fax-problem-statement-00.txt
>>  >
>>  > Paul its certainly not off topic and a reasonable idea, but not
>>  > practical
>>  > for a number of reasons. What you describe is a classic T.37 mode
>>  for
>>  > fax
>>  > that the old IETF Fax WG documented in RFC 3965. The problem is "non
>>  > repudiation". Fax has a certain legal standing and there is the
>>  > requirement
>>  > that there be actual documentation that a fax was sent by foo at
>>  such
>>  > and
>>  > such a time to bar and that the service provider or appropriate
>>  > intermediary
>>  > can provide the appropriate documentation of such a transaction.
>>  >
>>  > The problem is becoming more acute due to a variety of regulations,
>>  > HIPPA
>>  > and USA SOX, real estate etal which require the transmission of a
>>  > signature
>>  > as part of a legal filing and no, digital signatures have not proven
>>  to
>>  > be
>>  > successful even though they have legal standing in most countries
>>  now.
>>  > Health care records cannot be sent by email for privacy reasons
>>  > especially
>>  > test results.
>>  >
>>  > Fax is pretty simple and it works for these class of transactions.
>>  As a
>>  > result my contacts in the fax industry have noted a minor resurgence
>>  in
>>  > fax
>>  > machine and fax port sales.
>>  >
>>  > Some of us years ago thought using a variant of the Internet Print
>>  > Protocol,
>>  > based on HTTP for real time document transmission would have been a
>>  > better
>>  > idea but that went nowhere as well.
>>  >
>>  > The issue is there something in SIP signaling that can alert the
>>  first
>>  > hop
>>  > what this session really is such as ;USER=FAX that can help. I don't
>>  > know,
>>  > that is what needs to be investigated.
>>  >
>>  > >  -----Original Message-----
>>  > >  From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>>  > >  Sent: Thursday, December 24, 2009 3:34 PM
>>  > >  To: Richard Shockey
>>  > >  Cc: 'Dean Willis'; 'Paul E. Jones'; dispatch@ietf.org;
>>  > >  sipcore@ietf.org
>>  > >  Subject: Re: [dispatch] [sipcore] FW: I-D Action:draft-jones-sip-
>>  > >  forum-fax-problem-statement-00.txt
>>  > >
>>  > >  I know this is a bit off topic, and doesn't alleviate the
>>  problem,
>>  > but
>>  > >  has anybody ever proposed that an INVITE with offer for fax be
>>  > >  responded
>>  > >    to with a 3xx containing a mailto: URI? Or that ENUM for a fax
>>  > >  device
>>  > >  yield a mailto uri?
>>  > >
>>  > >  	Thanks,
>>  > >  	Paul
>>  > >
>>  > >  >
>>  > >  > Richard Shockey
>>  > >  > Shockey Consulting
>>  > >  > Chairman of the Board of Directors SIP Forum
>>  > >  > PSTN Mobile: +1 703.593.2683
>>  > >  > <mailto:richard(at)shockey.us>
>>  > >  > skype: rshockey101
>>  > >  > LinkedIn : http://www.linkedin.com/in/rshockey101
>>  > >  > http//www.sipforum.org
> 
> 
> 

From dean.willis@softarmor.com  Mon Dec 28 23:22:54 2009
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From: Dean Willis <dean.willis@softarmor.com>
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Cc: "'Paul E. Jones'" <paulej@packetizer.com>, dispatch@ietf.org, sipcore@ietf.org
Subject: Re: [sipcore] [dispatch] FW:	I-D	Action:draft-jones-sip-forum-fax-problem-statement-00.txt
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Paul Kyzivat wrote:
> I know this is a bit off topic, and doesn't alleviate the problem, but 
> has anybody ever proposed that an INVITE with offer for fax be responded 
>  to with a 3xx containing a mailto: URI? Or that ENUM for a fax device 
> yield a mailto uri?

I was going to say "redirect to T.37, since real-time fax over VoIP is 
just a bad idea" but Paul beat me to it.

Seriously: fax over IP really only works when the real-time conversion 
is done close to the edge of the IP network. and the core transit is 
store-and-forward/ The close to the consumer edge, the better. Trying to 
retrofit the timing equirements of fax onto RTP is an exercise in 
futility for those who don't understand the concept of lossy networks.

--
Dean



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Subject: [sipcore] FYI: ATT to FCC POTS is Dead..
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http://gigaom.com/2009/12/30/att-to-fcc-let-my-landlines-go/

Nice work folks..

Happy New Year!

