
From peter.musgrave@magorcorp.com  Sat May  1 03:51:02 2010
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Date: Sat, 1 May 2010 06:50:43 -0400
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From: Peter Musgrave <peter.musgrave@magorcorp.com>
To: Adam Roach <adam@estacado.net>
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Subject: Re: [sipcore] Call for Consensus: INFO Specification and Congestion
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Works for me (and happens to be what my application does).

Peter Musgrave

On Fri, Apr 30, 2010 at 5:59 PM, Adam Roach <adam@estacado.net> wrote:
> [as chair]
>
> During IESG review of the INFO specification, several members expressed
> concerns about the congestion properties of INFO. The draft authors and I=
ESG
> worked together to craft text that satisfies these concerns.
>
> Since the resulting text contains normative language, we are bringing the
> proposal to the SIPCORE working group for a 2-week consensus call. Please
> comment on the following proposed text before Friday, May 14th. If you ag=
ree
> with the proposal, please send a short response indicating that fact.
>
> "10.9. =A0Rate and size of INFO Requests
>
> =A0 =A0"INFO messages differ from many other sorts of SIP messages in tha=
t
> =A0 =A0 they carry application information, and the size and rate of the
> =A0 =A0 INFO message is directly determined by the application. This can
> =A0 =A0 cause application information traffic to interfere with other tra=
ffic
> =A0 =A0 on that infrastructure, or to self-interfere when data rates beco=
me
> =A0 =A0 too high.
>
> =A0 =A0"A designer of an INFO package, and the application that uses it,
> =A0 =A0 need to consider the impact that the size and the rate of the INF=
O
> =A0 =A0 messages have on the network and on other traffic, since it norma=
lly
> =A0 =A0 cannot be ensured that INFO messages will be carried over a
> =A0 =A0 congestion-controlled transport protocol end-to-end. Even if an
> =A0 =A0 INFO message is sent over a such a transport protocol, a downstre=
am
> =A0 =A0 SIP entity might forward the message over a transport protocol th=
at
> =A0 =A0 does not provide congestion control.
>
> =A0 =A0"Therefore, an application SHOULD generally have at most one outst=
anding
> INFO message.
> =A0 =A0 The guidelines in RFC3261 Section 18 MUST be followed for sending=
 and
> receiving INFO
> =A0 =A0 requests and responses, such as the guideline to not exceed messa=
ges
> =A0 =A0 sizes of 1300 bytes when the path MTU is unknown. RFC5405 provide=
s
> =A0 =A0 additional guidelines for applications using the User Datagram
> =A0 =A0 Protocol (UDP) that may be useful background reading."
>
> Thanks.
>
> /a
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>

From christer.holmberg@ericsson.com  Sat May  1 05:46:09 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adam Roach <adam@estacado.net>, SIPCORE <sipcore@ietf.org>
Date: Sat, 1 May 2010 14:32:59 +0200
Thread-Topic: [sipcore] Call for Consensus: INFO Specification and Congestion
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Subject: Re: [sipcore] Call for Consensus: INFO Specification and Congestion
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"INFO package" will of course be changed to "Info Package", in order to be =
alligned with the terminology in the rest of the document.

Regards,

Christer
________________________________________
From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On Behalf Of Adam=
 Roach [adam@estacado.net]
Sent: Saturday, May 01, 2010 12:59 AM
To: SIPCORE
Subject: [sipcore] Call for Consensus: INFO Specification and Congestion

[as chair]

During IESG review of the INFO specification, several members expressed
concerns about the congestion properties of INFO. The draft authors and
IESG worked together to craft text that satisfies these concerns.

Since the resulting text contains normative language, we are bringing
the proposal to the SIPCORE working group for a 2-week consensus call.
Please comment on the following proposed text before Friday, May 14th.
If you agree with the proposal, please send a short response indicating
that fact.

"10.9.  Rate and size of INFO Requests

     "INFO messages differ from many other sorts of SIP messages in that
      they carry application information, and the size and rate of the
      INFO message is directly determined by the application. This can
      cause application information traffic to interfere with other traffic
      on that infrastructure, or to self-interfere when data rates become
      too high.

     "A designer of an INFO package, and the application that uses it,
      need to consider the impact that the size and the rate of the INFO
      messages have on the network and on other traffic, since it normally
      cannot be ensured that INFO messages will be carried over a
      congestion-controlled transport protocol end-to-end. Even if an
      INFO message is sent over a such a transport protocol, a downstream
      SIP entity might forward the message over a transport protocol that
      does not provide congestion control.

     "Therefore, an application SHOULD generally have at most one
outstanding INFO message.
      The guidelines in RFC3261 Section 18 MUST be followed for sending
and receiving INFO
      requests and responses, such as the guideline to not exceed messages
      sizes of 1300 bytes when the path MTU is unknown. RFC5405 provides
      additional guidelines for applications using the User Datagram
      Protocol (UDP) that may be useful background reading."

Thanks.

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

From brett@broadsoft.com  Sat May  1 13:00:00 2010
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From: Brett Tate <brett@broadsoft.com>
To: Adam Roach <adam@estacado.net>, SIPCORE <sipcore@ietf.org>
Date: Sat, 1 May 2010 12:58:46 -0700
Thread-Topic: [sipcore] Call for Consensus: INFO Specification and Congestion
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Within proposed section 10.9 paragraph 2, "over a such a transport" should =
be "over such a transport".

The following sentence from paragraph 2 potentially could be improved.  How=
ever it is likely good enough since I'm not currently motivated enough to p=
ropose alternative wording.

"The guidelines in RFC3261 Section 18 MUST be followed for sending and rece=
iving INFO requests and responses, such as the guideline to not exceed mess=
ages sizes of 1300 bytes when the path MTU is unknown."

Concerning the above sentence... since the INFO draft is built upon RFC3261=
, there is little reason to indicate that the guidelines in RFC3261 Section=
 18 MUST be followed; the draft is not upgrading/adjusting any of the norma=
tive and non normative text from section 18.  Concerning the referenced gui=
deline, section 18 also has an additional guideline conflicting with the re=
ference guideline (i.e. upon failure to successfully switch away from UDP, =
the large request SHOULD still be sent over UDP).

Thanks,
Brett


From keith.drage@alcatel-lucent.com  Sat May  1 17:38:33 2010
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Adam Roach <adam@estacado.net>, SIPCORE <sipcore@ietf.org>
Date: Sun, 2 May 2010 02:36:51 +0200
Thread-Topic: [sipcore] Call for Consensus: INFO Specification and Congestion
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Subject: Re: [sipcore] Call for Consensus: INFO Specification and Congestion
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I have significant issues with the third proposed paragraph.

>      "Therefore, an application SHOULD generally have at most=20
> one outstanding INFO message.

This sentence does not solve the problem of congestion, which is the ethos =
for adding the text. All it does is impose limitations on implementations t=
hat are unnecessary. I would claim that I can impose congestion on the netw=
ork by poor design of INFO packages even without this limitation, whereas c=
areful use of INFO with multiple overlapping sends does not cause congestio=
n. Therefore I do not accept this sentence.

>       The guidelines in RFC3261 Section 18 MUST be followed=20
> for sending and receiving INFO
>       requests and responses, such as the guideline to not=20
> exceed messages
>       sizes of 1300 bytes when the path MTU is unknown.=20
> RFC5405 provides
>       additional guidelines for applications using the User Datagram
>       Protocol (UDP) that may be useful background reading."

The MUST here is superfluous. By all means draw our attention to the text i=
n RFC 3261, which already contains a MUST. But if people are really going t=
o ignore RFC 3261, then they will also ignore the MUST here. There are lots=
 of other MUSTs in RFC 3261 it is equally important for an INFO implementat=
ion to follow, and we get nowhere by repeating them all. I would suggest th=
at rather than the text here, you merely informatively introduce a quote fr=
om RFC 3261 of the relevant text.



regards

Keith=20


> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Adam Roach
> Sent: Friday, April 30, 2010 10:59 PM
> To: SIPCORE
> Subject: [sipcore] Call for Consensus: INFO Specification and=20
> Congestion
>=20
> [as chair]
>=20
> During IESG review of the INFO specification, several members=20
> expressed concerns about the congestion properties of INFO.=20
> The draft authors and IESG worked together to craft text that=20
> satisfies these concerns.
>=20
> Since the resulting text contains normative language, we are=20
> bringing the proposal to the SIPCORE working group for a=20
> 2-week consensus call.=20
> Please comment on the following proposed text before Friday,=20
> May 14th.=20
> If you agree with the proposal, please send a short response=20
> indicating that fact.
>=20
> "10.9.  Rate and size of INFO Requests
>=20
>      "INFO messages differ from many other sorts of SIP=20
> messages in that
>       they carry application information, and the size and rate of the
>       INFO message is directly determined by the application. This can
>       cause application information traffic to interfere with=20
> other traffic
>       on that infrastructure, or to self-interfere when data=20
> rates become
>       too high.
>=20
>      "A designer of an INFO package, and the application that uses it,
>       need to consider the impact that the size and the rate=20
> of the INFO
>       messages have on the network and on other traffic,=20
> since it normally
>       cannot be ensured that INFO messages will be carried over a
>       congestion-controlled transport protocol end-to-end. Even if an
>       INFO message is sent over a such a transport protocol,=20
> a downstream
>       SIP entity might forward the message over a transport=20
> protocol that
>       does not provide congestion control.
>=20
>      "Therefore, an application SHOULD generally have at most=20
> one outstanding INFO message.
>       The guidelines in RFC3261 Section 18 MUST be followed=20
> for sending and receiving INFO
>       requests and responses, such as the guideline to not=20
> exceed messages
>       sizes of 1300 bytes when the path MTU is unknown.=20
> RFC5405 provides
>       additional guidelines for applications using the User Datagram
>       Protocol (UDP) that may be useful background reading."
>=20
> Thanks.
>=20
> /a
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =

From christer.holmberg@ericsson.com  Sun May  2 01:00:24 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Brett Tate <brett@broadsoft.com>, Adam Roach <adam@estacado.net>, SIPCORE <sipcore@ietf.org>
Date: Sun, 2 May 2010 09:57:50 +0200
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Subject: Re: [sipcore] Call for Consensus: INFO Specification and Congestion
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Hi,

I am adding Lars Eggert to the discussion, since he had most issues with th=
is in IESG.

>Within proposed section 10.9 paragraph 2, "over a such a transport" should=
 be "over such a transport".

Correct.

>The following sentence from paragraph 2 potentially could be improved.  Ho=
wever it is likely good enough since I'm not currently motivated enough to =
propose alternative wording.
>
>"The guidelines in RFC3261 Section 18 MUST be followed for sending and rec=
eiving INFO requests and responses, such as the guideline to not exceed mes=
sages sizes of 1300 bytes when the path MTU is unknown."
>
>Concerning the above sentence... since the INFO draft is built upon RFC326=
1, there is little reason to indicate that the guidelines in RFC3261 Sectio=
n 18 MUST be followed; the draft is not upgrading/adjusting any of the norm=
ative and non normative text from section 18. =20
>Concerning the referenced guideline, section 18 also has an additional gui=
deline conflicting with the reference guideline (i.e. upon failure to succe=
ssfully switch away from UDP, the large request SHOULD still be sent over U=
DP).

Keith had similar comments, so I suggest we discuss it in that thread.

Thanks!

Regards,

Christer=

From christer.holmberg@ericsson.com  Sun May  2 01:03:39 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>, Adam Roach <adam@estacado.net>, SIPCORE <sipcore@ietf.org>
Date: Sun, 2 May 2010 10:00:14 +0200
Thread-Topic: [sipcore] Call for Consensus: INFO Specification and Congestion
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Cc: "lars.eggert@nokia.com" <lars.eggert@nokia.com>
Subject: Re: [sipcore] Call for Consensus: INFO Specification and Congestion
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Hi Keith,

I am adding Lars Eggert to the discussion, since he had most issues with th=
is in IESG.

>I have significant issues with the third proposed paragraph.
>
>"Therefore, an application SHOULD generally have at most one outstanding I=
NFO message.
>
>This sentence does not solve the problem of congestion, which is the ethos=
 for adding the text. All it does is >impose limitations on implementations=
 that are unnecessary. I would claim that I can impose congestion on the >n=
etwork by poor design of INFO packages even without this limitation, wherea=
s careful use of INFO with >multiple overlapping sends does not cause conge=
stion. Therefore I do not accept this sentence.

I'll let Lars comment on this.

>>The guidelines in RFC3261 Section 18 MUST be followed
>>for sending and receiving INFO requests and responses, such as the guidel=
ine to not
>>exceed messages sizes of 1300 bytes when the path MTU is unknown.
>>RFC5405 provides additional guidelines for applications using the User Da=
tagram
>>Protocol (UDP) that may be useful background reading."
>
>The MUST here is superfluous. By all means draw our attention to the text =
in RFC 3261, which already=20
>contains a MUST. But if people are really going to ignore RFC 3261, then t=
hey will also ignore the MUST=20
>here. There are lots of other MUSTs in RFC 3261 it is equally important fo=
r an INFO implementation to=20
>follow, and we get nowhere by repeating them all. I would suggest that rat=
her than the text here, you merely=20
>informatively introduce a quote from RFC 3261 of the relevant text.

Would you be ok if we change "Section 18 MUST be followed" to "Section 18 a=
re to be followed"?

Regards,

Christer




> -----Original Message-----
> From: sipcore-bounces@ietf.org
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Adam Roach
> Sent: Friday, April 30, 2010 10:59 PM
> To: SIPCORE
> Subject: [sipcore] Call for Consensus: INFO Specification and
> Congestion
>
> [as chair]
>
> During IESG review of the INFO specification, several members
> expressed concerns about the congestion properties of INFO.
> The draft authors and IESG worked together to craft text that
> satisfies these concerns.
>
> Since the resulting text contains normative language, we are
> bringing the proposal to the SIPCORE working group for a
> 2-week consensus call.
> Please comment on the following proposed text before Friday,
> May 14th.
> If you agree with the proposal, please send a short response
> indicating that fact.
>
> "10.9.  Rate and size of INFO Requests
>
>      "INFO messages differ from many other sorts of SIP
> messages in that
>       they carry application information, and the size and rate of the
>       INFO message is directly determined by the application. This can
>       cause application information traffic to interfere with
> other traffic
>       on that infrastructure, or to self-interfere when data
> rates become
>       too high.
>
>      "A designer of an INFO package, and the application that uses it,
>       need to consider the impact that the size and the rate
> of the INFO
>       messages have on the network and on other traffic,
> since it normally
>       cannot be ensured that INFO messages will be carried over a
>       congestion-controlled transport protocol end-to-end. Even if an
>       INFO message is sent over a such a transport protocol,
> a downstream
>       SIP entity might forward the message over a transport
> protocol that
>       does not provide congestion control.
>
>      "Therefore, an application SHOULD generally have at most
> one outstanding INFO message.
>       The guidelines in RFC3261 Section 18 MUST be followed
> for sending and receiving INFO
>       requests and responses, such as the guideline to not
> exceed messages
>       sizes of 1300 bytes when the path MTU is unknown.
> RFC5405 provides
>       additional guidelines for applications using the User Datagram
>       Protocol (UDP) that may be useful background reading."
>
> 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 keith.drage@alcatel-lucent.com  Sun May  2 04:58:36 2010
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Adam Roach <adam@estacado.net>, SIPCORE <sipcore@ietf.org>
Date: Sun, 2 May 2010 13:57:42 +0200
Thread-Topic: [sipcore] Call for Consensus: INFO Specification and Congestion
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Cc: "lars.eggert@nokia.com" <lars.eggert@nokia.com>
Subject: Re: [sipcore] Call for Consensus: INFO Specification and Congestion
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> >>The guidelines in RFC3261 Section 18 MUST be followed for=20
> sending and=20
> >>receiving INFO requests and responses, such as the guideline to not=20
> >>exceed messages sizes of 1300 bytes when the path MTU is unknown.
> >>RFC5405 provides additional guidelines for applications=20
> using the User=20
> >>Datagram Protocol (UDP) that may be useful background reading."
> >
> >The MUST here is superfluous. By all means draw our attention to the=20
> >text in RFC 3261, which already contains a MUST. But if people are=20
> >really going to ignore RFC 3261, then they will also ignore the MUST=20
> >here. There are lots of other MUSTs in RFC 3261 it is=20
> equally important=20
> >for an INFO implementation to follow, and we get nowhere by=20
> repeating them all. I would suggest that rather than the text=20
> here, you merely informatively introduce a quote from RFC=20
> 3261 of the relevant text.
>=20
> Would you be ok if we change "Section 18 MUST be followed" to=20
> "Section 18 are to be followed"?

That at least removes the double specification of the same MUST in two docu=
ments one of which is already dependent on the other. However I would still=
 point out that there are lots of RFC 3261 MUSTs that I could break in an I=
NFO implementation that would cause equal impact on the network. Therefore =
I still consider it unnecessary.

Can I further point out however that none of this text does not belong in s=
ection 10, as section 10 is about what needs to be documented to define an =
INFO package, not about what the implementor actually needs to do. It actua=
lly relates to whether it was appropriate to select INFO in the first place=
 to do the job, as if the information transfer requirements are inappropria=
te to the underlying transport capabilities of SIP, then INFO should not ha=
ve been selected in the first place.

regards

Keith=20

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: Sunday, May 02, 2010 9:00 AM
> To: DRAGE, Keith (Keith); Adam Roach; SIPCORE
> Cc: lars.eggert@nokia.com
> Subject: RE: [sipcore] Call for Consensus: INFO Specification=20
> and Congestion
>=20
> Hi Keith,
>=20
> I am adding Lars Eggert to the discussion, since he had most=20
> issues with this in IESG.
>=20
> >I have significant issues with the third proposed paragraph.
> >
> >"Therefore, an application SHOULD generally have at most one=20
> outstanding INFO message.
> >
> >This sentence does not solve the problem of congestion,=20
> which is the ethos for adding the text. All it does is=20
> >impose limitations on implementations that are unnecessary.=20
> I would claim that I can impose congestion on the >network by=20
> poor design of INFO packages even without this limitation,=20
> whereas careful use of INFO with >multiple overlapping sends=20
> does not cause congestion. Therefore I do not accept this sentence.
>=20
> I'll let Lars comment on this.
>=20
> >>The guidelines in RFC3261 Section 18 MUST be followed for=20
> sending and=20
> >>receiving INFO requests and responses, such as the guideline to not=20
> >>exceed messages sizes of 1300 bytes when the path MTU is unknown.
> >>RFC5405 provides additional guidelines for applications=20
> using the User=20
> >>Datagram Protocol (UDP) that may be useful background reading."
> >
> >The MUST here is superfluous. By all means draw our attention to the=20
> >text in RFC 3261, which already contains a MUST. But if people are=20
> >really going to ignore RFC 3261, then they will also ignore the MUST=20
> >here. There are lots of other MUSTs in RFC 3261 it is=20
> equally important=20
> >for an INFO implementation to follow, and we get nowhere by=20
> repeating them all. I would suggest that rather than the text=20
> here, you merely informatively introduce a quote from RFC=20
> 3261 of the relevant text.
>=20
> Would you be ok if we change "Section 18 MUST be followed" to=20
> "Section 18 are to be followed"?
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
>=20
> > -----Original Message-----
> > From: sipcore-bounces@ietf.org
> > [mailto:sipcore-bounces@ietf.org] On Behalf Of Adam Roach
> > Sent: Friday, April 30, 2010 10:59 PM
> > To: SIPCORE
> > Subject: [sipcore] Call for Consensus: INFO Specification and=20
> > Congestion
> >
> > [as chair]
> >
> > During IESG review of the INFO specification, several members=20
> > expressed concerns about the congestion properties of INFO.
> > The draft authors and IESG worked together to craft text that=20
> > satisfies these concerns.
> >
> > Since the resulting text contains normative language, we=20
> are bringing=20
> > the proposal to the SIPCORE working group for a 2-week=20
> consensus call.
> > Please comment on the following proposed text before=20
> Friday, May 14th.
> > If you agree with the proposal, please send a short response=20
> > indicating that fact.
> >
> > "10.9.  Rate and size of INFO Requests
> >
> >      "INFO messages differ from many other sorts of SIP messages in=20
> > that
> >       they carry application information, and the size and=20
> rate of the
> >       INFO message is directly determined by the=20
> application. This can
> >       cause application information traffic to interfere with other=20
> > traffic
> >       on that infrastructure, or to self-interfere when data rates=20
> > become
> >       too high.
> >
> >      "A designer of an INFO package, and the application=20
> that uses it,
> >       need to consider the impact that the size and the rate of the=20
> > INFO
> >       messages have on the network and on other traffic, since it=20
> > normally
> >       cannot be ensured that INFO messages will be carried over a
> >       congestion-controlled transport protocol end-to-end.=20
> Even if an
> >       INFO message is sent over a such a transport protocol, a=20
> > downstream
> >       SIP entity might forward the message over a transport=20
> protocol=20
> > that
> >       does not provide congestion control.
> >
> >      "Therefore, an application SHOULD generally have at most one=20
> > outstanding INFO message.
> >       The guidelines in RFC3261 Section 18 MUST be followed for=20
> > sending and receiving INFO
> >       requests and responses, such as the guideline to not exceed=20
> > messages
> >       sizes of 1300 bytes when the path MTU is unknown.
> > RFC5405 provides
> >       additional guidelines for applications using the User Datagram
> >       Protocol (UDP) that may be useful background reading."
> >
> > 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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Subject: [sipcore] Review request: Dial String syntax for the "tel" URI Scheme
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All,

=20

This is a request for the list to review an Internet Draft that codifies
the parameters for Dial Strings to be used in an rfc3966 compliant URL.
The requirements for this draft originated in the IEEE/ISTO Printer
Working Group (PWG) <http://www.pwg.org>.  The PWG is a consortium
printer and multi-function device manufacturers, print server
developers, operating system providers and print management application
developers.  Previous works of the PWG include the Printer MIB (rfc3805)
and the Internet Printing Protocol (rfc2911, rfc2910 and others)  The
Working Group in the PWG that I chair is addressing the modeling of
Multifunction Devices.  In that effort an outbound facsimile (i.e.
FaxOut) requires a way to encode Dial Strings in the URL for a
destination facsimile device.

=20

The draft title is "Dial String syntax for the "tel" URI Scheme" and it
can be retrieved from
<http://tools.ietf.org/html/draft-zehler-teldial-info-00>.   It proposes
three "tel" URI schemes parameters that cover pre-dialing, post-dialing
and T.33 subaddressing.  The post-dialing and T.33 subaddressing are
adopted from the now obsolete rfc2806 but updated for tfc3966 syntax.
Pre-dial was created to avoid obfuscation of the resource identified by
the URI.  Rfc2806 allowed the pre-dial string to be included in the tel:
URI.

=20

In a comment on this Internet Draft on the URI Review list it was
suggested that the RAI-area working groups that deals with SIP should be
the place where the Dial String parameter semantics should be discussed.


=20

I look forward to your comments,

=20

Peter Zehler




Xerox Research Center Webster
Email: Peter.Zehler@Xerox.com
Voice: (585) 265-8755
FAX: (585) 265-7441
US Mail: Peter Zehler
Xerox Corp.
800 Phillips Rd.
M/S 128-25E
Webster NY, 14580-9701=20

=20

=20

>=20

> -----Original Message-----

> From: IETF I-D Submission Tool [mailto:idsubmission@ietf.org]

> Sent: Monday, April 26, 2010 4:06 PM

> To: Zehler, Peter

> Subject: New Version Notification for draft-zehler-teldial-info-00

>=20

>=20

> A new version of I-D, draft-zehler-teldial-info-00.txt has been
successfully submitted by Peter Zehler and posted to the IETF
repository.

>=20

> Filename:        draft-zehler-teldial-info

> Revision:        00

> Title:           Dial String syntax for the "tel" URI Scheme

> Creation_date:   2010-04-24

> WG ID:           Independent Submission

> Number_of_pages: 7

>=20

> Abstract:

> The "tel" URI (Uniform Resource Identifier) for Telephone Numbers=20

> [RFC3966] describes resources identified by telephone numbers.  These=20

> resources specify termination points within a public or private=20

> telephone network. The termination points are a globally or locally=20

> unique endpoint.  RFC3966 does not address the means by which the=20

> telephone URI can be used to reach termination point from a specific=20

> node in the telephone network.

>=20

> This document specifies the means by which the termination point is=20

> reached from a specific node in the telephone network. This document=20

> specifies how to encode the numbers, symbols and pauses required to=20

> traverse the phone network to reach the termination point in an RFC

> 3966 compliant "tel" URI.

>=20

>=20

>=20

> The IETF Secretariat.

>=20

>=20

=20

=20

=20

Peter Zehler

Xerox Research Center Webster
Email: Peter.Zehler@Xerox.com <mailto:Peter.Zehler@Xerox.com>=20
Voice: (585) 265-8755
FAX: (585) 265-7441
US Mail: Peter Zehler
Xerox Corp.
800 Phillips Rd.
M/S 128-25E
Webster NY, 14580-9701=20

=20


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<div class=3DSection1>

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

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

<p class=3DMsoNormal>This is a request for the list to review an =
Internet Draft
that codifies the parameters for Dial Strings to be used in an rfc3966
compliant URL.&nbsp; The requirements for this draft originated in the
IEEE/ISTO Printer Working Group (PWG) &lt;<a =
href=3D"http://www.pwg.org">http://www.pwg.org</a>&gt;.&nbsp;
The PWG is a consortium <span style=3D'color:black'>printer and =
multi-function
device manufacturers, print server developers, operating system =
providers and
print management application developers.</span>&nbsp; Previous works of =
the PWG
include the Printer MIB (rfc3805) and the Internet Printing Protocol =
(rfc2911,
rfc2910 and others)&nbsp; The Working Group in the PWG that I chair is
addressing the modeling of Multifunction Devices.&nbsp; In that effort =
an
outbound facsimile (i.e. FaxOut) requires a way to encode Dial Strings =
in the
URL for a destination facsimile device.<o:p></o:p></p>

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

<p class=3DMsoNormal>The draft title is &quot;Dial String syntax for the
&quot;tel&quot; URI Scheme&quot; and it can be retrieved from &lt;<a
href=3D"http://tools.ietf.org/html/draft-zehler-teldial-info-00">http://t=
ools.ietf.org/html/draft-zehler-teldial-info-00</a>&gt;.&nbsp;
&nbsp;It proposes three &#8220;tel&#8221; URI schemes parameters that =
cover
pre-dialing, post-dialing &nbsp;and T.33 subaddressing.&nbsp; The =
post-dialing
and T.33 subaddressing are adopted from the now obsolete rfc2806 but =
updated
for tfc3966 syntax.&nbsp; Pre-dial was created to avoid obfuscation of =
the
resource identified by the URI.&nbsp; Rfc2806 allowed the pre-dial =
string to be
included in the tel: URI.<o:p></o:p></p>

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

<p class=3DMsoNormal>In a comment on this Internet Draft on the URI =
Review list
it was suggested that the RAI-area working groups that deals with SIP =
should be
the place where the Dial String parameter semantics should be =
discussed.&nbsp; <o:p></o:p></p>

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

<p class=3DMsoNormal>I look forward to your comments,<o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoPlainText><span =
style=3D'font-family:"Impact","sans-serif";
color:navy'>Peter Zehler</span><span style=3D'color:#1F497D'><br>
<br>
</span><o:p></o:p></p>

<p class=3DMsoPlainText><span style=3D'color:#1F497D'><br>
</span><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";
color:navy'>Xerox Research Center Webster<br>
</span><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>Email: <a =
href=3D"mailto:Peter.Zehler@Xerox.com">Peter.Zehler@Xerox.com</a></span><=
span
style=3D'color:#1F497D'><br>
</span><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>Voice: (585) 265-8755</span><span =
style=3D'color:#1F497D'><br>
</span><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>FAX: (585) 265-7441</span><span style=3D'color:#1F497D'><br>
</span><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>US Mail: Peter Zehler</span><span =
style=3D'color:#1F497D'><br>
</span><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>Xerox Corp.</span><span style=3D'color:#1F497D'><br>
</span><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>800 Phillips Rd.</span><span style=3D'color:#1F497D'><br>
</span><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>M/S 128-25E</span><span style=3D'color:#1F497D'><br>
</span><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>Webster NY, 14580-9701</span><span style=3D'color:#1F497D'> =
</span><span
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif";color:#1F497D'><o:p></o:p></span></p>

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

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

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

<p class=3DMsoPlainText>&gt; -----Original Message-----<o:p></o:p></p>

<p class=3DMsoPlainText>&gt; From: IETF I-D Submission Tool =
[mailto:idsubmission@ietf.org]<o:p></o:p></p>

<p class=3DMsoPlainText>&gt; Sent: Monday, April 26, 2010 4:06 =
PM<o:p></o:p></p>

<p class=3DMsoPlainText>&gt; To: Zehler, Peter<o:p></o:p></p>

<p class=3DMsoPlainText>&gt; Subject: New Version Notification for
draft-zehler-teldial-info-00<o:p></o:p></p>

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

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

<p class=3DMsoPlainText>&gt; A new version of I-D,
draft-zehler-teldial-info-00.txt has been successfully submitted by =
Peter Zehler
and posted to the IETF repository.<o:p></o:p></p>

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

<p class=3DMsoPlainText>&gt; Filename: &nbsp; &nbsp; &nbsp;
&nbsp;draft-zehler-teldial-info<o:p></o:p></p>

<p class=3DMsoPlainText>&gt; Revision: &nbsp; &nbsp; &nbsp; =
&nbsp;00<o:p></o:p></p>

<p class=3DMsoPlainText>&gt; Title: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
Dial
String syntax for the &quot;tel&quot; URI Scheme<o:p></o:p></p>

<p class=3DMsoPlainText>&gt; Creation_date: &nbsp; =
2010-04-24<o:p></o:p></p>

<p class=3DMsoPlainText>&gt; WG ID: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
Independent Submission<o:p></o:p></p>

<p class=3DMsoPlainText>&gt; Number_of_pages: 7<o:p></o:p></p>

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

<p class=3DMsoPlainText>&gt; Abstract:<o:p></o:p></p>

<p class=3DMsoPlainText>&gt; The &quot;tel&quot; URI (Uniform Resource
Identifier) for Telephone Numbers <o:p></o:p></p>

<p class=3DMsoPlainText>&gt; [RFC3966] describes resources identified by
telephone numbers. &nbsp;These <o:p></o:p></p>

<p class=3DMsoPlainText>&gt; resources specify termination points within =
a public
or private <o:p></o:p></p>

<p class=3DMsoPlainText>&gt; telephone network. The termination points =
are a
globally or locally <o:p></o:p></p>

<p class=3DMsoPlainText>&gt; unique endpoint. &nbsp;RFC3966 does not =
address the
means by which the <o:p></o:p></p>

<p class=3DMsoPlainText>&gt; telephone URI can be used to reach =
termination point
from a specific <o:p></o:p></p>

<p class=3DMsoPlainText>&gt; node in the telephone =
network.<o:p></o:p></p>

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

<p class=3DMsoPlainText>&gt; This document specifies the means by which =
the
termination point is <o:p></o:p></p>

<p class=3DMsoPlainText>&gt; reached from a specific node in the =
telephone
network. This document <o:p></o:p></p>

<p class=3DMsoPlainText>&gt; specifies how to encode the numbers, =
symbols and
pauses required to <o:p></o:p></p>

<p class=3DMsoPlainText>&gt; traverse the phone network to reach the =
termination
point in an RFC<o:p></o:p></p>

<p class=3DMsoPlainText>&gt; 3966 compliant &quot;tel&quot; =
URI.<o:p></o:p></p>

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

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

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

<p class=3DMsoPlainText>&gt; The IETF Secretariat.<o:p></o:p></p>

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

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

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

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

<p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-family:"Impact","sans-serif";color:navy'>Peter
Zehler</span><span style=3D'color:#1F497D'><br>
<br>
</span><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";
color:navy'>Xerox Research Center Webster<br>
</span><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>Email: <a href=3D"mailto:Peter.Zehler@Xerox.com"><span
style=3D'color:blue'>Peter.Zehler@Xerox.com</span></a></span><span
style=3D'color:#1F497D'><br>
</span><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>Voice: (585) 265-8755</span><span =
style=3D'color:#1F497D'><br>
</span><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>FAX: (585) 265-7441</span><span style=3D'color:#1F497D'><br>
</span><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>US Mail: Peter Zehler</span><span =
style=3D'color:#1F497D'><br>
</span><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>Xerox Corp.</span><span style=3D'color:#1F497D'><br>
</span><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>800 Phillips Rd.</span><span style=3D'color:#1F497D'><br>
</span><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>M/S 128-25E</span><span style=3D'color:#1F497D'><br>
</span><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:navy'>Webster NY, 14580-9701</span><span style=3D'color:#1F497D'> =
</span><span
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif";color:#1F497D'><o:p></o:p></span></p>

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

</div>

</body>

</html>

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From pkyzivat@cisco.com  Mon May  3 09:29:59 2010
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Cc: sipcore@ietf.org
Subject: Re: [sipcore] Review request: Dial String syntax for the "tel" URI Scheme
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[with my individual hat on]

Peter,

I find this to be a *terrible* idea!!!

This conflates the notion of a Uniform Resource *Identifier*, with that 
of a Uniform Resource *Locator*.

Were such changes made, it would complicate the use of tel: URIs in SIP, 
and even of sip: URIs, since sip incorporates the telephone-subscriber 
syntax into its own syntax.

For instance, this would confuse URI comparison. Do two URIs that differ 
only with regards to these new parameters compare as equal? For some 
purposes they should, but for others not.

Also, if you have a URI containing these parameters in your address 
book, and share it with me, are those parameters valid for me as well? 
(Most likely I should strip them out, and then perhaps add others in 
their place.)

In sip we eventually defined RFC 4967 to define a way to specify a dial 
string that is distinct from a tel: uri.

If such things are important, I suggest that they belong in a new URI 
scheme. (Or some existing one different from tel:)

	Thanks,
	Paul

P.S. I know there have been other parameters defined for tel: that have 
some of the same issues. IMO that was a mistake. Lets not do it again.

Zehler, Peter wrote:
> All,
> 
>  
> 
> This is a request for the list to review an Internet Draft that codifies 
> the parameters for Dial Strings to be used in an rfc3966 compliant URL.  
> The requirements for this draft originated in the IEEE/ISTO Printer 
> Working Group (PWG) <http://www.pwg.org>.  The PWG is a consortium 
> printer and multi-function device manufacturers, print server 
> developers, operating system providers and print management application 
> developers.  Previous works of the PWG include the Printer MIB (rfc3805) 
> and the Internet Printing Protocol (rfc2911, rfc2910 and others)  The 
> Working Group in the PWG that I chair is addressing the modeling of 
> Multifunction Devices.  In that effort an outbound facsimile (i.e. 
> FaxOut) requires a way to encode Dial Strings in the URL for a 
> destination facsimile device.
> 
>  
> 
> The draft title is "Dial String syntax for the "tel" URI Scheme" and it 
> can be retrieved from 
> <http://tools.ietf.org/html/draft-zehler-teldial-info-00>.   It proposes 
> three “tel” URI schemes parameters that cover pre-dialing, post-dialing 
>  and T.33 subaddressing.  The post-dialing and T.33 subaddressing are 
> adopted from the now obsolete rfc2806 but updated for tfc3966 syntax.  
> Pre-dial was created to avoid obfuscation of the resource identified by 
> the URI.  Rfc2806 allowed the pre-dial string to be included in the tel: 
> URI.
> 
>  
> 
> In a comment on this Internet Draft on the URI Review list it was 
> suggested that the RAI-area working groups that deals with SIP should be 
> the place where the Dial String parameter semantics should be discussed. 
> 
>  
> 
> I look forward to your comments,
> 
>  
> 
> Peter Zehler
> 
> 
> Xerox Research Center Webster
> Email: Peter.Zehler@Xerox.com <mailto:Peter.Zehler@Xerox.com>
> Voice: (585) 265-8755
> FAX: (585) 265-7441
> US Mail: Peter Zehler
> Xerox Corp.
> 800 Phillips Rd.
> M/S 128-25E
> Webster NY, 14580-9701
> 
>  
> 
>  
> 
>  > 
> 
>  > -----Original Message-----
> 
>  > From: IETF I-D Submission Tool [mailto:idsubmission@ietf.org]
> 
>  > Sent: Monday, April 26, 2010 4:06 PM
> 
>  > To: Zehler, Peter
> 
>  > Subject: New Version Notification for draft-zehler-teldial-info-00
> 
>  > 
> 
>  > 
> 
>  > A new version of I-D, draft-zehler-teldial-info-00.txt has been 
> successfully submitted by Peter Zehler and posted to the IETF repository.
> 
>  > 
> 
>  > Filename:        draft-zehler-teldial-info
> 
>  > Revision:        00
> 
>  > Title:           Dial String syntax for the "tel" URI Scheme
> 
>  > Creation_date:   2010-04-24
> 
>  > WG ID:           Independent Submission
> 
>  > Number_of_pages: 7
> 
>  > 
> 
>  > Abstract:
> 
>  > The "tel" URI (Uniform Resource Identifier) for Telephone Numbers
> 
>  > [RFC3966] describes resources identified by telephone numbers.  These
> 
>  > resources specify termination points within a public or private
> 
>  > telephone network. The termination points are a globally or locally
> 
>  > unique endpoint.  RFC3966 does not address the means by which the
> 
>  > telephone URI can be used to reach termination point from a specific
> 
>  > node in the telephone network.
> 
>  > 
> 
>  > This document specifies the means by which the termination point is
> 
>  > reached from a specific node in the telephone network. This document
> 
>  > specifies how to encode the numbers, symbols and pauses required to
> 
>  > traverse the phone network to reach the termination point in an RFC
> 
>  > 3966 compliant "tel" URI.
> 
>  > 
> 
>  > 
> 
>  > 
> 
>  > The IETF Secretariat.
> 
>  > 
> 
>  > 
> 
>  
> 
>  
> 
>  
> 
> Peter Zehler
> 
> Xerox Research Center Webster
> Email: Peter.Zehler@Xerox.com <mailto:Peter.Zehler@Xerox.com>
> Voice: (585) 265-8755
> FAX: (585) 265-7441
> US Mail: Peter Zehler
> Xerox Corp.
> 800 Phillips Rd.
> M/S 128-25E
> Webster NY, 14580-9701
> 
>  
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore

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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: "Zehler, Peter" <Peter.Zehler@xerox.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Mon, 3 May 2010 13:58:05 -0400
Thread-Topic: [sipcore] Review request: Dial String syntax for the "tel" URI Scheme
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________________________________________
From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On Behalf Of Zehl=
er, Peter [Peter.Zehler@xerox.com]

The draft title is "Dial String syntax for the "tel" URI Scheme" and it can=
 be retrieved from <http://tools.ietf.org/html/draft-zehler-teldial-info-00=
>.   It proposes three =93tel=94 URI schemes parameters that cover pre-dial=
ing, post-dialing  and T.33 subaddressing.  The post-dialing and T.33 subad=
dressing are adopted from the now obsolete rfc2806 but updated for tfc3966 =
syntax.  Pre-dial was created to avoid obfuscation of the resource identifi=
ed by the URI.  Rfc2806 allowed the pre-dial string to be included in the t=
el: URI.
________________________________________

In regard to subaddressing, this is a good idea, since the subaddressed des=
tination is a descrete entity with a name that is unique within the context=
 of the E.164 number.

In regard to "dial strings", I think this is not a good idea.  A "dial stri=
ng" is the specific routing from one location to one destination.  It isn't=
 a name for the destination, and it really isn't a URL, as there is the imp=
lication that a URL is globally valid.  What you are attempting to do is cr=
eate a URL-like string which is the dialing instructions to reach a particu=
lar destination from a particular origin and which also makes explicit the =
E.164 address of the destination.  It has no portability at all.

Also, the commonest transformation that is applied to convert E.164 numbers=
 to dial strings is the removal of the country and city codes from the begi=
nning of the E.164 number, and the draft has no mechanism for specifying th=
at.

I think a better approach would be to see if the context concept of RFC 396=
6 solves the problem, that is, the device which is acting on the tel URI sh=
ould be provisioned with an appropriate converstion table from telephone nu=
mbers to dialing actions.

If you really need to transport a string of dialing instructions from one e=
lement to another, it would be much safer to define a new namespace.  E.g.,=
 "dial:w9w01118005551212".

Dale

From fluffy@cisco.com  Tue May  4 11:54:51 2010
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Subject: [sipcore] Option tag problem ...
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There's been lots of discussion around changing the option tag =
registration process but I'm unclear on what the problem is we are =
trying to solve. Are there some concrete examples folks can bring up =
form the past where we would have a better interoperability situations =
today if we changed this? I'm not saying there is not a problem, I just =
think we need to get clear about what we are trying to fix before we are =
going to be able to figure out how to fix it.=20



Cullen Jennings
For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html



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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Cullen Jennings'" <fluffy@cisco.com>
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Cc: 'Michael Miller' <mlm.michael.miller@gmail.com>, sipcore@ietf.org, 'Hadriel Kaplan' <HKaplan@acmepacket.com>
Subject: Re: [sipcore] SIP OPTIONS "ping"
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Cullen,

Problem Statement:

Recognizing that if a SIP entity does not know the basic operational state
of a peer device, there is a risk of delays in delivering service and
weakened ability for administrators to correct service issues.

Further recognizing that SIP specifies the use of OPTIONS as a means of
determining the basic state of a UAS, but RFC 3261's description of the
usage is shallow and does not lend itself to interoperability, resulting in
a wide variety of implementations of OPTIONS for this purpose. 

Proposed Solution:

Define procedures for using the current OPTIONS method defined in RFC 3261
in a way that improves interoperability and lends to consistent
implementation when used to determine the operational status of peer SIP
entities so as to reduce delays, improve overall operational efficiency, and
provide reliable/useful operational status information to administrators.

Paul

> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf Of Cullen Jennings
> Sent: Friday, April 30, 2010 3:21 PM
> To: Paul E. Jones
> Cc: 'Michael Miller'; sipcore@ietf.org; 'Hadriel Kaplan'
> Subject: Re: [sipcore] SIP OPTIONS "ping"
> 
> 
> Thanks for the reset.
> 
> I think the reasons this thread has been so unproductive is we are
> unclear on what the problem is we are trying to solve. "We need a way
> to do PING" is not a problem. It is a solution to some problem. If the
> problem is figure out which of multiple next hops to use based on load
> balancing and if the hop is dead or alive, well the PING approach here
> will fail miserably (see the overload discussion and simulations) and
> we have WG going to the IESG to try and solve that problem.
> 
> Let me make it very clear I am against the working group working on
> this draft until someone can describe what the problem is. The current
> draft hand waves at a few problems - does not say what it is trying to
> do. Then proposes some syntax with no backing semantics that would help
> solve any of the problems it mentions.
> 
> Cullen
> 
> On Apr 29, 2010, at 7:11 PM, Paul E. Jones wrote:
> 
> > Guys,
> >
> > I decided to start this new thread, because the previous one is
> proving to
> > be a bit challenging to respond to.
> >
> > Let's back up and review where we are and what we're trying to
> accomplish.
> >
> > Many devices today use OPTIONS as a "ping" method.  The reason is
> that there
> > was nothing better or more appropriate for an out-of-dialog means of
> > determining status of a peer SIP device.  These devices include
> PBXes, SBCs,
> > etc.
> >
> > The problems are:
> > 1) Response codes vary from one device to another (e.g., 600 vs. 503
> vs.
> > 486)
> > 2) Nowhere is it documented that an OPTIONS can be used as an
> > application-level "ping"
> >
> > Taking note of the current state of affairs and desiring to improve
> it
> > without changing SIP in any significant way, we prepared the OPTIONS
> "ping"
> > draft.
> >
> > Is it a ping message or not?  The thread has focused heavily on the
> contents
> > of the message, but we're not proposing changes to the contents of a
> > message.  We only wish to focus on response codes: the contents of
> headers
> > and body were not subject to change.
> >
> > We proposed sending OPTIONS "ping" messages to the IP address of the
> peer
> > device with a request URI of "sip:hostport" since that seemed most
> logical
> > to us.  Christer pointed out that that's the address used when
> addressing a
> > SIP proxy server.  I think that's OK, since we're not suggesting that
> the
> > proxy server actually alter its behavior.  That said, we would like
> to
> > propose use of 503 to indicate that the device is overloaded, rather
> than
> > 600: that's what RFC 3261 says. But, note that we're not saying a 600
> is
> > illegal if that were the appropriate code -- it just isn't in this
> case.
> >
> > We also expect this to be used only between two devices that agree to
> do so.
> > The interval for messages I also wanted to keep locally configured.
> >
> > It might be useful to indicate that the message was a "ping" -- I
> have no
> > objection.  But, I would not implement a different code path,
> personally.
> > That's why I felt it wasn't terribly important to include.
> >
> > If we do nothing, the world will continue to use OPTIONS as they do
> today.
> > Introducing PING will take time to standardize and we still have to
> support
> > OPTIONS for a decade.  So, why bother?
> >
> > My recommendation is that we not boil the ocean and try to create the
> most
> > architecturally pure solution.  Let's recognize the use of OPTIONS in
> this
> > way as legitimate, keep it peer-to-peer (maxforward=1), recommend
> response
> > codes, and expect a locally provisioned usage policy (e.g., message
> > frequency).
> >
> > I've captured the groups' comments in this draft, though I'd like to
> > reinstate the paragraph relating to the SIP Request URI
> (sip:hostport):
> > http://hive.packetizer.com/users/paulej/internet-drafts/pre-draft-
> jones-sip-
> > options-ping-01.pdf
> >
> > In any case, I'll go along with whatever the group wishes to do.
> >
> > Paul
> >
> >
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
> 
> 
> Cullen Jennings
> For corporate legal information go to:
> http://www.cisco.com/web/about/doing_business/legal/cri/index.html
> 
> 
> 
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 



From Madhubabu@veraznet.com  Tue May  4 17:49:28 2010
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From: Madhubabu Brahmanapally <Madhubabu@veraznet.com>
To: "sipcore@ietf.org" <sipcore@ietf.org>
Date: Tue, 4 May 2010 17:49:13 -0700
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Hi,
Are there any work/draft available for updates of RFC5407 for Race conditio=
ns or best practices?

Regards
Madhubabu

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<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
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font-family:Arial'>Hi,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>Are there any work/draft available for updates of RFC540=
7
for Race conditions or best practices?<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
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<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>Regards<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
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From paulej@packetizer.com  Tue May  4 21:49:39 2010
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Subject: [sipcore] draft-jones-sip-options-ping-01
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Folks,

I posted a revised version of the OPTIONS "ping" draft with changes as per
our discussions.  One change I made was that I re-instated the paragraph
related to addressing, saying OPTIONS "should" be addressed to sip:hostport.
I'm still willing to consider removing that entirely if that's preferred,
but I recall that Hadriel preferred to have something that one could
generally assume to be the proper address.  As a means of querying UAS
status, I think that's a reasonable choice.

Anyway, I welcome comments on the draft.

Paul



From christer.holmberg@ericsson.com  Tue May  4 22:29:01 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Wed, 5 May 2010 07:20:19 +0200
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Hi,

As said earlier, 3261 already says that OPTIONS addressed to proxies are to=
 use sip:hostport, so I don't think we need to say anything about that.

Regards,

Christer=20

> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Paul E. Jones
> Sent: 5. toukokuuta 2010 7:49
> To: sipcore@ietf.org
> Subject: [sipcore] draft-jones-sip-options-ping-01
>=20
> Folks,
>=20
> I posted a revised version of the OPTIONS "ping" draft with=20
> changes as per our discussions.  One change I made was that I=20
> re-instated the paragraph related to addressing, saying=20
> OPTIONS "should" be addressed to sip:hostport.
> I'm still willing to consider removing that entirely if=20
> that's preferred, but I recall that Hadriel preferred to have=20
> something that one could generally assume to be the proper=20
> address.  As a means of querying UAS status, I think that's a=20
> reasonable choice.
>=20
> Anyway, I welcome comments on the draft.
>=20
> Paul
>=20
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =

From john.elwell@siemens-enterprise.com  Wed May  5 05:55:02 2010
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: "Paul E. Jones" <paulej@packetizer.com>, 'Cullen Jennings' <fluffy@cisco.com>
Date: Wed, 5 May 2010 14:54:41 +0200
Thread-Topic: [sipcore] SIP OPTIONS "ping"
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Cc: 'Michael Miller' <mlm.michael.miller@gmail.com>, "sipcore@ietf.org" <sipcore@ietf.org>, 'Hadriel Kaplan' <HKaplan@acmepacket.com>
Subject: Re: [sipcore] SIP OPTIONS "ping"
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=20

> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Paul E. Jones
> Sent: 05 May 2010 00:37
> To: 'Cullen Jennings'
> Cc: 'Michael Miller'; sipcore@ietf.org; 'Hadriel Kaplan'
> Subject: Re: [sipcore] SIP OPTIONS "ping"
>=20
> Cullen,
>=20
> Problem Statement:
>=20
> Recognizing that if a SIP entity does not know the basic=20
> operational state
> of a peer device, there is a risk of delays in delivering service and
> weakened ability for administrators to correct service issues.
[JRE] To a first approximation, a 2xx versus a 4xx/5xx/6xx should tell you =
the basic operational state of the responder. So we need more precise infor=
mation on what is broken.

John


>=20
> Further recognizing that SIP specifies the use of OPTIONS as=20
> a means of
> determining the basic state of a UAS, but RFC 3261's=20
> description of the
> usage is shallow and does not lend itself to=20
> interoperability, resulting in
> a wide variety of implementations of OPTIONS for this purpose.=20
>=20
> Proposed Solution:
>=20
> Define procedures for using the current OPTIONS method=20
> defined in RFC 3261
> in a way that improves interoperability and lends to consistent
> implementation when used to determine the operational status=20
> of peer SIP
> entities so as to reduce delays, improve overall operational=20
> efficiency, and
> provide reliable/useful operational status information to=20
> administrators.
>=20
> Paul
>=20
> > -----Original Message-----
> > From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> > Behalf Of Cullen Jennings
> > Sent: Friday, April 30, 2010 3:21 PM
> > To: Paul E. Jones
> > Cc: 'Michael Miller'; sipcore@ietf.org; 'Hadriel Kaplan'
> > Subject: Re: [sipcore] SIP OPTIONS "ping"
> >=20
> >=20
> > Thanks for the reset.
> >=20
> > I think the reasons this thread has been so unproductive is we are
> > unclear on what the problem is we are trying to solve. "We=20
> need a way
> > to do PING" is not a problem. It is a solution to some=20
> problem. If the
> > problem is figure out which of multiple next hops to use=20
> based on load
> > balancing and if the hop is dead or alive, well the PING=20
> approach here
> > will fail miserably (see the overload discussion and=20
> simulations) and
> > we have WG going to the IESG to try and solve that problem.
> >=20
> > Let me make it very clear I am against the working group working on
> > this draft until someone can describe what the problem is.=20
> The current
> > draft hand waves at a few problems - does not say what it=20
> is trying to
> > do. Then proposes some syntax with no backing semantics=20
> that would help
> > solve any of the problems it mentions.
> >=20
> > Cullen
> >=20
> > On Apr 29, 2010, at 7:11 PM, Paul E. Jones wrote:
> >=20
> > > Guys,
> > >
> > > I decided to start this new thread, because the previous one is
> > proving to
> > > be a bit challenging to respond to.
> > >
> > > Let's back up and review where we are and what we're trying to
> > accomplish.
> > >
> > > Many devices today use OPTIONS as a "ping" method.  The reason is
> > that there
> > > was nothing better or more appropriate for an=20
> out-of-dialog means of
> > > determining status of a peer SIP device.  These devices include
> > PBXes, SBCs,
> > > etc.
> > >
> > > The problems are:
> > > 1) Response codes vary from one device to another (e.g.,=20
> 600 vs. 503
> > vs.
> > > 486)
> > > 2) Nowhere is it documented that an OPTIONS can be used as an
> > > application-level "ping"
> > >
> > > Taking note of the current state of affairs and desiring=20
> to improve
> > it
> > > without changing SIP in any significant way, we prepared=20
> the OPTIONS
> > "ping"
> > > draft.
> > >
> > > Is it a ping message or not?  The thread has focused=20
> heavily on the
> > contents
> > > of the message, but we're not proposing changes to the=20
> contents of a
> > > message.  We only wish to focus on response codes: the contents of
> > headers
> > > and body were not subject to change.
> > >
> > > We proposed sending OPTIONS "ping" messages to the IP=20
> address of the
> > peer
> > > device with a request URI of "sip:hostport" since that seemed most
> > logical
> > > to us.  Christer pointed out that that's the address used when
> > addressing a
> > > SIP proxy server.  I think that's OK, since we're not=20
> suggesting that
> > the
> > > proxy server actually alter its behavior.  That said, we=20
> would like
> > to
> > > propose use of 503 to indicate that the device is=20
> overloaded, rather
> > than
> > > 600: that's what RFC 3261 says. But, note that we're not=20
> saying a 600
> > is
> > > illegal if that were the appropriate code -- it just isn't in this
> > case.
> > >
> > > We also expect this to be used only between two devices=20
> that agree to
> > do so.
> > > The interval for messages I also wanted to keep locally=20
> configured.
> > >
> > > It might be useful to indicate that the message was a "ping" -- I
> > have no
> > > objection.  But, I would not implement a different code path,
> > personally.
> > > That's why I felt it wasn't terribly important to include.
> > >
> > > If we do nothing, the world will continue to use OPTIONS=20
> as they do
> > today.
> > > Introducing PING will take time to standardize and we=20
> still have to
> > support
> > > OPTIONS for a decade.  So, why bother?
> > >
> > > My recommendation is that we not boil the ocean and try=20
> to create the
> > most
> > > architecturally pure solution.  Let's recognize the use=20
> of OPTIONS in
> > this
> > > way as legitimate, keep it peer-to-peer (maxforward=3D1), recommend
> > response
> > > codes, and expect a locally provisioned usage policy=20
> (e.g., message
> > > frequency).
> > >
> > > I've captured the groups' comments in this draft, though=20
> I'd like to
> > > reinstate the paragraph relating to the SIP Request URI
> > (sip:hostport):
> > > http://hive.packetizer.com/users/paulej/internet-drafts/pre-draft-
> > jones-sip-
> > > options-ping-01.pdf
> > >
> > > In any case, I'll go along with whatever the group wishes to do.
> > >
> > > Paul
> > >
> > >
> > > _______________________________________________
> > > sipcore mailing list
> > > sipcore@ietf.org
> > > https://www.ietf.org/mailman/listinfo/sipcore
> >=20
> >=20
> > Cullen Jennings
> > For corporate legal information go to:
> > http://www.cisco.com/web/about/doing_business/legal/cri/index.html
> >=20
> >=20
> >=20
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
> >=20
>=20
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =

From christer.holmberg@ericsson.com  Wed May  5 06:28:58 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>, Adam Roach <adam@estacado.net>, SIPCORE <sipcore@ietf.org>
Date: Wed, 5 May 2010 15:28:39 +0200
Thread-Topic: [sipcore] Call for Consensus: INFO Specification and Congestion
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Cc: "lars.eggert@nokia.com" <lars.eggert@nokia.com>
Subject: Re: [sipcore] Call for Consensus: INFO Specification and Congestion
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Hi,=20

>>>>The guidelines in RFC3261 Section 18 MUST be followed for sending and
>>>>receiving INFO requests and responses, such as the guideline to not=20
>>>>exceed messages sizes of 1300 bytes when the path MTU is unknown.
>>>>RFC5405 provides additional guidelines for applications using the User
>>>>Datagram Protocol (UDP) that may be useful background reading."
>>>
>>>The MUST here is superfluous. By all means draw our attention to the=20
>>>text in RFC 3261, which already contains a MUST. But if people are=20
>>>really going to ignore RFC 3261, then they will also ignore the MUST=20
>>>here. There are lots of other MUSTs in RFC 3261 it is equally important
>>>for an INFO implementation to follow, and we get nowhere by repeating th=
em all.=20
>>>I would suggest that rather than the text here, you merely informatively=
=20
>>>introduce a quote from RFC 3261 of the relevant text.
>>=20
>>Would you be ok if we change "Section 18 MUST be followed" to "Section=20
>>18 are to be followed"?
>=20
>That at least removes the double specification of the same=20
>MUST in two documents one of which is already dependent on=20
>the other. However I would still point out that there are=20
>lots of RFC 3261 MUSTs that I could break in an INFO=20
>implementation that would cause equal impact on the network.=20
>Therefore I still consider it unnecessary.
>=20
>Can I further point out however that none of this text does=20
>not belong in section 10, as section 10 is about what needs=20
>to be documented to define an INFO package, not about what=20
>the implementor actually needs to do. It actually relates to=20
>whether it was appropriate to select INFO in the first place=20
>to do the job, as if the information transfer requirements=20
>are inappropriate to the underlying transport capabilities of=20
>SIP, then INFO should not have been selected in the first place.

So, I guess it should be somewhere in section 7.4.

Regards,

Christer




=20
> > -----Original Message-----
> > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > Sent: Sunday, May 02, 2010 9:00 AM
> > To: DRAGE, Keith (Keith); Adam Roach; SIPCORE
> > Cc: lars.eggert@nokia.com
> > Subject: RE: [sipcore] Call for Consensus: INFO Specification and=20
> > Congestion
> >=20
> > Hi Keith,
> >=20
> > I am adding Lars Eggert to the discussion, since he had most issues=20
> > with this in IESG.
> >=20
> > >I have significant issues with the third proposed paragraph.
> > >
> > >"Therefore, an application SHOULD generally have at most one
> > outstanding INFO message.
> > >
> > >This sentence does not solve the problem of congestion,
> > which is the ethos for adding the text. All it does is
> > >impose limitations on implementations that are unnecessary.=20
> > I would claim that I can impose congestion on the >network by poor=20
> > design of INFO packages even without this limitation,=20
> whereas careful=20
> > use of INFO with >multiple overlapping sends does not cause=20
> > congestion. Therefore I do not accept this sentence.
> >=20
> > I'll let Lars comment on this.
> >=20
> > >>The guidelines in RFC3261 Section 18 MUST be followed for
> > sending and
> > >>receiving INFO requests and responses, such as the=20
> guideline to not=20
> > >>exceed messages sizes of 1300 bytes when the path MTU is unknown.
> > >>RFC5405 provides additional guidelines for applications
> > using the User
> > >>Datagram Protocol (UDP) that may be useful background reading."
> > >
> > >The MUST here is superfluous. By all means draw our=20
> attention to the=20
> > >text in RFC 3261, which already contains a MUST. But if people are=20
> > >really going to ignore RFC 3261, then they will also=20
> ignore the MUST=20
> > >here. There are lots of other MUSTs in RFC 3261 it is
> > equally important
> > >for an INFO implementation to follow, and we get nowhere by
> > repeating them all. I would suggest that rather than the text here,=20
> > you merely informatively introduce a quote from RFC
> > 3261 of the relevant text.
> >=20
> > Would you be ok if we change "Section 18 MUST be followed"=20
> to "Section=20
> > 18 are to be followed"?
> >=20
> > Regards,
> >=20
> > Christer
> >=20
> >=20
> >=20
> >=20
> > > -----Original Message-----
> > > From: sipcore-bounces@ietf.org
> > > [mailto:sipcore-bounces@ietf.org] On Behalf Of Adam Roach
> > > Sent: Friday, April 30, 2010 10:59 PM
> > > To: SIPCORE
> > > Subject: [sipcore] Call for Consensus: INFO Specification and=20
> > > Congestion
> > >
> > > [as chair]
> > >
> > > During IESG review of the INFO specification, several members=20
> > > expressed concerns about the congestion properties of INFO.
> > > The draft authors and IESG worked together to craft text that=20
> > > satisfies these concerns.
> > >
> > > Since the resulting text contains normative language, we
> > are bringing
> > > the proposal to the SIPCORE working group for a 2-week
> > consensus call.
> > > Please comment on the following proposed text before
> > Friday, May 14th.
> > > If you agree with the proposal, please send a short response=20
> > > indicating that fact.
> > >
> > > "10.9.  Rate and size of INFO Requests
> > >
> > >      "INFO messages differ from many other sorts of SIP=20
> messages in=20
> > > that
> > >       they carry application information, and the size and
> > rate of the
> > >       INFO message is directly determined by the
> > application. This can
> > >       cause application information traffic to interfere=20
> with other=20
> > > traffic
> > >       on that infrastructure, or to self-interfere when=20
> data rates=20
> > > become
> > >       too high.
> > >
> > >      "A designer of an INFO package, and the application
> > that uses it,
> > >       need to consider the impact that the size and the=20
> rate of the=20
> > > INFO
> > >       messages have on the network and on other traffic, since it=20
> > > normally
> > >       cannot be ensured that INFO messages will be carried over a
> > >       congestion-controlled transport protocol end-to-end.=20
> > Even if an
> > >       INFO message is sent over a such a transport protocol, a=20
> > > downstream
> > >       SIP entity might forward the message over a transport
> > protocol
> > > that
> > >       does not provide congestion control.
> > >
> > >      "Therefore, an application SHOULD generally have at most one=20
> > > outstanding INFO message.
> > >       The guidelines in RFC3261 Section 18 MUST be followed for=20
> > > sending and receiving INFO
> > >       requests and responses, such as the guideline to not exceed=20
> > > messages
> > >       sizes of 1300 bytes when the path MTU is unknown.
> > > RFC5405 provides
> > >       additional guidelines for applications using the=20
> User Datagram
> > >       Protocol (UDP) that may be useful background reading."
> > >
> > > 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 keith.drage@alcatel-lucent.com  Wed May  5 06:46:11 2010
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Adam Roach <adam@estacado.net>, SIPCORE <sipcore@ietf.org>
Date: Wed, 5 May 2010 15:45:21 +0200
Thread-Topic: [sipcore] Call for Consensus: INFO Specification and Congestion
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Cc: "lars.eggert@nokia.com" <lars.eggert@nokia.com>
Subject: Re: [sipcore] Call for Consensus: INFO Specification and Congestion
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Yes, but obviously not in the same words. It needs to be tailored as do not=
 choose an INFO solution if you need to do this.

Keith =20

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: Wednesday, May 05, 2010 2:29 PM
> To: DRAGE, Keith (Keith); Adam Roach; SIPCORE
> Cc: lars.eggert@nokia.com
> Subject: RE: [sipcore] Call for Consensus: INFO Specification=20
> and Congestion
>=20
>=20
> Hi,=20
>=20
> >>>>The guidelines in RFC3261 Section 18 MUST be followed for sending=20
> >>>>and receiving INFO requests and responses, such as the=20
> guideline to=20
> >>>>not exceed messages sizes of 1300 bytes when the path MTU=20
> is unknown.
> >>>>RFC5405 provides additional guidelines for applications using the=20
> >>>>User Datagram Protocol (UDP) that may be useful=20
> background reading."
> >>>
> >>>The MUST here is superfluous. By all means draw our=20
> attention to the=20
> >>>text in RFC 3261, which already contains a MUST. But if people are=20
> >>>really going to ignore RFC 3261, then they will also=20
> ignore the MUST=20
> >>>here. There are lots of other MUSTs in RFC 3261 it is equally=20
> >>>important for an INFO implementation to follow, and we get=20
> nowhere by repeating them all.
> >>>I would suggest that rather than the text here, you merely=20
> >>>informatively introduce a quote from RFC 3261 of the relevant text.
> >>=20
> >>Would you be ok if we change "Section 18 MUST be followed"=20
> to "Section
> >>18 are to be followed"?
> >=20
> >That at least removes the double specification of the same=20
> MUST in two=20
> >documents one of which is already dependent on the other. However I=20
> >would still point out that there are lots of RFC 3261 MUSTs that I=20
> >could break in an INFO implementation that would cause equal=20
> impact on=20
> >the network.
> >Therefore I still consider it unnecessary.
> >=20
> >Can I further point out however that none of this text does=20
> not belong=20
> >in section 10, as section 10 is about what needs to be documented to=20
> >define an INFO package, not about what the implementor=20
> actually needs=20
> >to do. It actually relates to whether it was appropriate to=20
> select INFO=20
> >in the first place to do the job, as if the information transfer=20
> >requirements are inappropriate to the underlying transport=20
> capabilities=20
> >of SIP, then INFO should not have been selected in the first place.
>=20
> So, I guess it should be somewhere in section 7.4.
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
>=20
> =20
> > > -----Original Message-----
> > > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > > Sent: Sunday, May 02, 2010 9:00 AM
> > > To: DRAGE, Keith (Keith); Adam Roach; SIPCORE
> > > Cc: lars.eggert@nokia.com
> > > Subject: RE: [sipcore] Call for Consensus: INFO Specification and=20
> > > Congestion
> > >=20
> > > Hi Keith,
> > >=20
> > > I am adding Lars Eggert to the discussion, since he had=20
> most issues=20
> > > with this in IESG.
> > >=20
> > > >I have significant issues with the third proposed paragraph.
> > > >
> > > >"Therefore, an application SHOULD generally have at most one
> > > outstanding INFO message.
> > > >
> > > >This sentence does not solve the problem of congestion,
> > > which is the ethos for adding the text. All it does is
> > > >impose limitations on implementations that are unnecessary.=20
> > > I would claim that I can impose congestion on the=20
> >network by poor=20
> > > design of INFO packages even without this limitation,
> > whereas careful
> > > use of INFO with >multiple overlapping sends does not cause=20
> > > congestion. Therefore I do not accept this sentence.
> > >=20
> > > I'll let Lars comment on this.
> > >=20
> > > >>The guidelines in RFC3261 Section 18 MUST be followed for
> > > sending and
> > > >>receiving INFO requests and responses, such as the
> > guideline to not
> > > >>exceed messages sizes of 1300 bytes when the path MTU=20
> is unknown.
> > > >>RFC5405 provides additional guidelines for applications
> > > using the User
> > > >>Datagram Protocol (UDP) that may be useful background reading."
> > > >
> > > >The MUST here is superfluous. By all means draw our
> > attention to the
> > > >text in RFC 3261, which already contains a MUST. But if=20
> people are=20
> > > >really going to ignore RFC 3261, then they will also
> > ignore the MUST
> > > >here. There are lots of other MUSTs in RFC 3261 it is
> > > equally important
> > > >for an INFO implementation to follow, and we get nowhere by
> > > repeating them all. I would suggest that rather than the=20
> text here,=20
> > > you merely informatively introduce a quote from RFC
> > > 3261 of the relevant text.
> > >=20
> > > Would you be ok if we change "Section 18 MUST be followed"=20
> > to "Section
> > > 18 are to be followed"?
> > >=20
> > > Regards,
> > >=20
> > > Christer
> > >=20
> > >=20
> > >=20
> > >=20
> > > > -----Original Message-----
> > > > From: sipcore-bounces@ietf.org
> > > > [mailto:sipcore-bounces@ietf.org] On Behalf Of Adam Roach
> > > > Sent: Friday, April 30, 2010 10:59 PM
> > > > To: SIPCORE
> > > > Subject: [sipcore] Call for Consensus: INFO Specification and=20
> > > > Congestion
> > > >
> > > > [as chair]
> > > >
> > > > During IESG review of the INFO specification, several members=20
> > > > expressed concerns about the congestion properties of INFO.
> > > > The draft authors and IESG worked together to craft text that=20
> > > > satisfies these concerns.
> > > >
> > > > Since the resulting text contains normative language, we
> > > are bringing
> > > > the proposal to the SIPCORE working group for a 2-week
> > > consensus call.
> > > > Please comment on the following proposed text before
> > > Friday, May 14th.
> > > > If you agree with the proposal, please send a short response=20
> > > > indicating that fact.
> > > >
> > > > "10.9.  Rate and size of INFO Requests
> > > >
> > > >      "INFO messages differ from many other sorts of SIP
> > messages in
> > > > that
> > > >       they carry application information, and the size and
> > > rate of the
> > > >       INFO message is directly determined by the
> > > application. This can
> > > >       cause application information traffic to interfere
> > with other
> > > > traffic
> > > >       on that infrastructure, or to self-interfere when
> > data rates
> > > > become
> > > >       too high.
> > > >
> > > >      "A designer of an INFO package, and the application
> > > that uses it,
> > > >       need to consider the impact that the size and the
> > rate of the
> > > > INFO
> > > >       messages have on the network and on other=20
> traffic, since it=20
> > > > normally
> > > >       cannot be ensured that INFO messages will be=20
> carried over a
> > > >       congestion-controlled transport protocol end-to-end.=20
> > > Even if an
> > > >       INFO message is sent over a such a transport protocol, a=20
> > > > downstream
> > > >       SIP entity might forward the message over a transport
> > > protocol
> > > > that
> > > >       does not provide congestion control.
> > > >
> > > >      "Therefore, an application SHOULD generally have=20
> at most one=20
> > > > outstanding INFO message.
> > > >       The guidelines in RFC3261 Section 18 MUST be followed for=20
> > > > sending and receiving INFO
> > > >       requests and responses, such as the guideline to=20
> not exceed=20
> > > > messages
> > > >       sizes of 1300 bytes when the path MTU is unknown.
> > > > RFC5405 provides
> > > >       additional guidelines for applications using the
> > User Datagram
> > > >       Protocol (UDP) that may be useful background reading."
> > > >
> > > > 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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Subject: Re: [sipcore] SIP OPTIONS "ping"
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On May 4, 2010, at 5:37 PM, Paul E. Jones wrote:

> Cullen,
>=20
> Problem Statement:
>=20
> Recognizing that if a SIP entity does not know the basic operational =
state
> of a peer device, there is a risk of delays in delivering service and
> weakened ability for administrators to correct service issues.

So for many devices, there is one one logical next hop and the existing =
DNS and ICMP approaches seem to be adequate but for other situations, =
there has always been a desire to do something better than that. So for =
the type of servers that had many possible next hops to choose from, I =
think people do want to have a better way to choose which next hop to =
use that minimizes delays due to servers being down or overloaded. Right =
now a new working group has been proposed to work on that problem. Is =
there anything you see we need to do to expand the scope of the SIP =
Overload Control Charter to make sure the problem you are envision is in =
scope for that WG? As far as I can tell, this is the simple case of what =
they hope to solve.=20

>=20
> Further recognizing that SIP specifies the use of OPTIONS as a means =
of
> determining the basic state of a UAS, but RFC 3261's description of =
the
> usage is shallow and does not lend itself to interoperability, =
resulting in
> a wide variety of implementations of OPTIONS for this purpose.=20
>=20


Actually, I don't think this was the purpose of OPTIONS in 3261 so I'm =
not surprised there are some difficulties in using it this way. However, =
I don't see that as a problem.=20


Cullen




Cullen Jennings
For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html




From wwwrun@rfc-editor.org  Wed May  5 17:53:54 2010
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Subject: [sipcore] RFC 5839 on An Extension to Session Initiation Protocol (SIP) Events for Conditional Event Notification
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A new Request for Comments is now available in online RFC libraries.

        
        RFC 5839

        Title:      An Extension to Session Initiation 
                    Protocol (SIP) Events for Conditional Event 
                    Notification 
        Author:     A. Niemi, D. Willis, Ed.
        Status:     Standards Track
        Stream:     IETF
        Date:       May 2010
        Mailbox:    aki.niemi@nokia.com, 
                    dean.willis@softarmor.com
        Pages:      25
        Characters: 52999
        Updates/Obsoletes/SeeAlso:   None

        I-D Tag:    draft-ietf-sipcore-subnot-etags-04.txt

        URL:        http://www.rfc-editor.org/rfc/rfc5839.txt

The Session Initiation Protocol (SIP) events framework enables
receiving asynchronous notification of various events from other SIP
user agents.  This framework defines the procedures for creating,
refreshing, and terminating subscriptions, as well as fetching and
periodic polling of resource state.  These procedures provide no
tools to avoid replaying event notifications that have already been
received by a user agent.  This memo defines an extension to SIP
events that allows the subscriber to condition the subscription
request to whether the state has changed since the previous
notification was received.  When such a condition is true, either the
body of a resulting event notification or the entire notification
message is suppressed.  [STANDARDS TRACK]

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

This is now a Proposed Standard Protocol.

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

This announcement is sent to the IETF-Announce and rfc-dist lists.
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The RFC Editor Team
Association Management Solutions, LLC



From gsalguei@cisco.com  Wed May  5 19:08:53 2010
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Subject: Re: [sipcore] draft-jones-sip-options-ping-01
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Christer -=20

I think we are in agreement here. Do you think the language should be =
revised to something stronger (i.e. MUST) in regards to using that =
address form when sending an OPTIONS to a proxy? Or do you think it =
should stay as is?

Gonzalo

=20
On May 5, 2010, at 1:20 AM, Christer Holmberg wrote:

>=20
> Hi,
>=20
> As said earlier, 3261 already says that OPTIONS addressed to proxies =
are to use sip:hostport, so I don't think we need to say anything about =
that.
>=20
> Regards,
>=20
> Christer=20
>=20
>> -----Original Message-----
>> From: sipcore-bounces@ietf.org=20
>> [mailto:sipcore-bounces@ietf.org] On Behalf Of Paul E. Jones
>> Sent: 5. toukokuuta 2010 7:49
>> To: sipcore@ietf.org
>> Subject: [sipcore] draft-jones-sip-options-ping-01
>>=20
>> Folks,
>>=20
>> I posted a revised version of the OPTIONS "ping" draft with=20
>> changes as per our discussions.  One change I made was that I=20
>> re-instated the paragraph related to addressing, saying=20
>> OPTIONS "should" be addressed to sip:hostport.
>> I'm still willing to consider removing that entirely if=20
>> that's preferred, but I recall that Hadriel preferred to have=20
>> something that one could generally assume to be the proper=20
>> address.  As a means of querying UAS status, I think that's a=20
>> reasonable choice.
>>=20
>> Anyway, I welcome comments on the draft.
>>=20
>> Paul
>>=20
>>=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 christer.holmberg@ericsson.com  Wed May  5 22:33:57 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Gonzalo Salgueiro <gsalguei@cisco.com>
Date: Thu, 6 May 2010 07:33:37 +0200
Thread-Topic: [sipcore] draft-jones-sip-options-ping-01
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Cc: "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] draft-jones-sip-options-ping-01
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Hi,=20

>I think we are in agreement here. Do you think the language=20
>should be revised to something stronger (i.e. MUST) in=20
>regards to using that address form when sending an OPTIONS to=20
>a proxy? Or do you think it should stay as is?

Similar to comments that have been given for INFO, we should not "duplicate=
" MUST/SHALL/SHOULD/etc that are already in RFC3261.

We can say something: "The addressing of OPTIONS requests to a proxy is des=
cribed in section X in RFC 3261"

...or something like that.

Regards,

Christer




> On May 5, 2010, at 1:20 AM, Christer Holmberg wrote:
>=20
> >=20
> > Hi,
> >=20
> > As said earlier, 3261 already says that OPTIONS addressed=20
> to proxies are to use sip:hostport, so I don't think we need=20
> to say anything about that.
> >=20
> > Regards,
> >=20
> > Christer
> >=20
> >> -----Original Message-----
> >> From: sipcore-bounces@ietf.org
> >> [mailto:sipcore-bounces@ietf.org] On Behalf Of Paul E. Jones
> >> Sent: 5. toukokuuta 2010 7:49
> >> To: sipcore@ietf.org
> >> Subject: [sipcore] draft-jones-sip-options-ping-01
> >>=20
> >> Folks,
> >>=20
> >> I posted a revised version of the OPTIONS "ping" draft=20
> with changes=20
> >> as per our discussions.  One change I made was that I=20
> re-instated the=20
> >> paragraph related to addressing, saying OPTIONS "should"=20
> be addressed=20
> >> to sip:hostport.
> >> I'm still willing to consider removing that entirely if that's=20
> >> preferred, but I recall that Hadriel preferred to have=20
> something that=20
> >> one could generally assume to be the proper address.  As a=20
> means of=20
> >> querying UAS status, I think that's a reasonable choice.
> >>=20
> >> Anyway, I welcome comments on the draft.
> >>=20
> >> Paul
> >>=20
> >>=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
>=20
> =

From HKaplan@acmepacket.com  Wed May  5 22:41:40 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Thu, 6 May 2010 01:40:46 -0400
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> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behal=
f
> Of Christer Holmberg
> Sent: Wednesday, May 05, 2010 1:20 AM
>=20
> As said earlier, 3261 already says that OPTIONS addressed to proxies are
> to use sip:hostport, so I don't think we need to say anything about that.

But this is *not* to a proxy, or at least not always to a proxy.  The draft=
 is for it between UA's as well (B2BUA's to be precise, but UA's nonetheles=
s).

-hadriel

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, "Paul E. Jones" <paulej@packetizer.com>, 'Cullen Jennings' <fluffy@cisco.com>
Date: Thu, 6 May 2010 01:55:05 -0400
Thread-Topic: [sipcore] SIP OPTIONS "ping"
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Subject: Re: [sipcore] SIP OPTIONS "ping"
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> -----Original Message-----
> From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]
> Sent: Wednesday, May 05, 2010 8:55 AM
> To: Paul E. Jones; 'Cullen Jennings'
>=20
> > -----Original Message-----
> > From: sipcore-bounces@ietf.org
> > [mailto:sipcore-bounces@ietf.org] On Behalf Of Paul E. Jones
> > Sent: 05 May 2010 00:37
> >
> > Problem Statement:
> > Recognizing that if a SIP entity does not know the basic
> > operational state
> > of a peer device, there is a risk of delays in delivering service and
> > weakened ability for administrators to correct service issues.
>
> [JRE] To a first approximation, a 2xx versus a 4xx/5xx/6xx should tell yo=
u
> the basic operational state of the responder. So we need more precise
> information on what is broken.

But it doesn't in practice.  Both because it's not a binary on/off operatio=
nal state, and because of targeting/addressing issues. =20

Some devices send a 503 when they're too busy, and don't want new requests =
for retry-after time, but still expect in-dialog requests.  Some devices se=
nd a 486 instead.

Some devices send a 503 when they're admin-down.  Some devices send a 500 i=
nstead.  In either case, some expect in-dialog messages, others don't.

Some send a 483 Too Many Hops, because the requestor had to set a max-forwa=
rds of 0 or 1 to prevent it going past the next-hop. (because the addressin=
g for the OPTIONS is ambiguous)  And that prevents either a 2xx or 5xx.

-hadriel

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To: Cullen Jennings <fluffy@cisco.com>, "Paul E. Jones" <paulej@packetizer.com>
Date: Thu, 6 May 2010 02:03:24 -0400
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Subject: Re: [sipcore] SIP OPTIONS "ping"
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> -----Original Message-----
> From: Cullen Jennings [mailto:fluffy@cisco.com]
> Sent: Wednesday, May 05, 2010 10:56 AM
> To: Paul E. Jones
>=20
> So for many devices, there is one one logical next hop and the existing
> DNS and ICMP approaches seem to be adequate but for other situations,
> there has always been a desire to do something better than that. So for
> the type of servers that had many possible next hops to choose from, I
> think people do want to have a better way to choose which next hop to use
> that minimizes delays due to servers being down or overloaded. Right now =
a
> new working group has been proposed to work on that problem. Is there
> anything you see we need to do to expand the scope of the SIP Overload
> Control Charter to make sure the problem you are envision is in scope for
> that WG? As far as I can tell, this is the simple case of what they hope
> to solve.

Yes, I also noted that when Paul first submitted the draft.  But his point =
was that (1) it's not just load state, it's also admin state, and (2) it's =
a ping as well.  I suppose one could argue that if the overload WG covers r=
eal state information and not just load, then OPTIONS could be the carrier =
for that info.


> > Further recognizing that SIP specifies the use of OPTIONS as a means of
> > determining the basic state of a UAS, but RFC 3261's description of the
> > usage is shallow and does not lend itself to interoperability, resultin=
g
> in
> > a wide variety of implementations of OPTIONS for this purpose.
>=20
> Actually, I don't think this was the purpose of OPTIONS in 3261 so I'm no=
t
> surprised there are some difficulties in using it this way. However, I
> don't see that as a problem.

>From rfc3261 page 68:

11.2 Processing of OPTIONS Request

   The response to an OPTIONS is constructed using the standard rules
   for a SIP response as discussed in Section 8.2.6.  The response code
   chosen MUST be the same that would have been chosen had the request
   been an INVITE.  That is, a 200 (OK) would be returned if the UAS is
   ready to accept a call, a 486 (Busy Here) would be returned if the
   UAS is busy, etc.  This allows an OPTIONS request to be used to
   determine the basic state of a UAS, which can be an indication of
   whether the UAS will accept an INVITE request.

-hadriel

From christer.holmberg@ericsson.com  Wed May  5 23:22:24 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>, "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Thu, 6 May 2010 08:22:05 +0200
Thread-Topic: [sipcore] draft-jones-sip-options-ping-01
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Subject: Re: [sipcore] draft-jones-sip-options-ping-01
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Hi,=20

>>As said earlier, 3261 already says that OPTIONS addressed to proxies=20
>>are to use sip:hostport, so I don't think we need to say anything about t=
hat.
>=20
>But this is *not* to a proxy, or at least not always to a=20
>proxy.  The draft is for it between UA's as well (B2BUA's to=20
>be precise, but UA's nonetheless).

True, I was also talking about proxies.

But, I believe 3261 also says that OPTIONS addresses to UAs are using the h=
ostpart. Would need to double check, though.

Regards,

Christer

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To: Christer Holmberg <christer.holmberg@ericsson.com>, "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>
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I don't think so - it was ambiguous for that.  It only said this in section=
 11:
   The target of the OPTIONS request is identified by the Request-URI,
   which could identify another UA or a SIP server.  If the OPTIONS is
   addressed to a proxy server, the Request-URI is set without a user
   part, similar to the way a Request-URI is set for a REGISTER request.

It didn't explicitly say how to address a UA instance, only a proxy. (of co=
urse how one knows a priori that the next-hop is a proxy vs. a b2bua is bey=
ond me) =20

But it doesn't really matter.  In practice, historically even using sip:hos=
tport has had problems, especially if the next-hop is an SBC.  Because an S=
BC doesn't know if the OPTIONS request is really for itself, vs. for an aut=
horitative proxy for the domain, behind it - if it knew the OPTIONS was for=
 a ping, it would know it's for itself; but if it's for capabilities then i=
t was always a guess as to whom it's for; and of course the OPTIONS doesn't=
 say which it is so it's always a guess.  So people set max-forwards to eit=
her 0 or 1 to avoid it going further, and that caused other issues (like ge=
tting a 483 instead of a 200, and incrementing loop-detection counters).

So today you have a bunch of configuration you may need to set, and to set =
it you need to know what the next-hop or previous-hop expects.  So it's err=
or-prone, tedious, and not fun.

-hadriel

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: Thursday, May 06, 2010 2:22 AM
> >
> >But this is *not* to a proxy, or at least not always to a
> >proxy.  The draft is for it between UA's as well (B2BUA's to
> >be precise, but UA's nonetheless).
>=20
> True, I was also talking about proxies.
>=20
> But, I believe 3261 also says that OPTIONS addresses to UAs are using the
> hostpart. Would need to double check, though.
>=20
> Regards,
>=20
> Christer

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Subject: Re: [sipcore] draft-jones-sip-options-ping-01
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If you dig into it, the notion of "addressing OPTIONS to a proxy" is not 
a meaningful concept, so defining rules for how to do it is meaningless.

This is because "proxy" is a *role* a server takes on, not a particular 
type of server. The same box may sometimes act like a proxy, and 
sometimes ack like something else. And by definition, when a box 
generates its own response to an incoming OPTIONS request, it is acting 
as a UAS.

The situation that is most proxy-like in this regard is when a request 
arrives, isn't addressed to the receiving server, and max-forwards is 
zero. If it returns 200 response in such a case, then I suppose we can 
conclude that it was at least a middle-box, though it might still 
normally act as a B2BUA rather than a proxy. In any case, in this 
situation, the request is by definition not addressed to this box, so 
imposing requirements on the form of the R-URI would be silly.

So fundamentally we need to just be talking about what it means to 
address a specific server, without regard to what sort of server it 
might be. And IMO that simply means using a URI with a domain that 
server "is responsible for". All URIs with that domain are in some sense 
addressed to that box, regardless of user part. At that point it is up 
to that box to interpret the user part as it sees fit. Some user parts 
may be treated by it as identifiers of certain resources within the box. 
Others might be treated as requiring retargeting. Note that even a URI 
with *no* user part could be treated as requiring retargeting. (E.g. the 
box might be a load balancer.)

	Thanks,
	Paul

	Thanks,
	Paul

Hadriel Kaplan wrote:
> I don't think so - it was ambiguous for that.  It only said this in section 11:
>    The target of the OPTIONS request is identified by the Request-URI,
>    which could identify another UA or a SIP server.  If the OPTIONS is
>    addressed to a proxy server, the Request-URI is set without a user
>    part, similar to the way a Request-URI is set for a REGISTER request.
> 
> It didn't explicitly say how to address a UA instance, only a proxy. (of course how one knows a priori that the next-hop is a proxy vs. a b2bua is beyond me)  
> 
> But it doesn't really matter.  In practice, historically even using sip:hostport has had problems, especially if the next-hop is an SBC.  Because an SBC doesn't know if the OPTIONS request is really for itself, vs. for an authoritative proxy for the domain, behind it - if it knew the OPTIONS was for a ping, it would know it's for itself; but if it's for capabilities then it was always a guess as to whom it's for; and of course the OPTIONS doesn't say which it is so it's always a guess.  So people set max-forwards to either 0 or 1 to avoid it going further, and that caused other issues (like getting a 483 instead of a 200, and incrementing loop-detection counters).
> 
> So today you have a bunch of configuration you may need to set, and to set it you need to know what the next-hop or previous-hop expects.  So it's error-prone, tedious, and not fun.
> 
> -hadriel
> 
>> -----Original Message-----
>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>> Sent: Thursday, May 06, 2010 2:22 AM
>>> But this is *not* to a proxy, or at least not always to a
>>> proxy.  The draft is for it between UA's as well (B2BUA's to
>>> be precise, but UA's nonetheless).
>> True, I was also talking about proxies.
>>
>> But, I believe 3261 also says that OPTIONS addresses to UAs are using the
>> hostpart. Would need to double check, though.
>>
>> Regards,
>>
>> Christer
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 

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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Christer Holmberg'" <christer.holmberg@ericsson.com>, "'Gonzalo Salgueiro'" <gsalguei@cisco.com>
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Subject: Re: [sipcore] draft-jones-sip-options-ping-01
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Christer,

Sounds good.  How is this?

    The OPTIONS message SHOULD be addressed to a peer SIP entity using
    a SIP URI of the form "sip:hostport" (see Section 25.1 of RFC 3261
    for definitions).  For example, "sip:192.168.1.5" is preferred, whereas
    "sip:bob@example.com" is not preferred for the purpose of this memo.
    The request URI required when addressing a proxy is described in
    Section 11 of RFC 3261.

Paul

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: Thursday, May 06, 2010 1:34 AM
> To: Gonzalo Salgueiro
> Cc: Paul E. Jones; sipcore@ietf.org
> Subject: RE: [sipcore] draft-jones-sip-options-ping-01
> 
> 
> Hi,
> 
> >I think we are in agreement here. Do you think the language
> >should be revised to something stronger (i.e. MUST) in
> >regards to using that address form when sending an OPTIONS to
> >a proxy? Or do you think it should stay as is?
> 
> Similar to comments that have been given for INFO, we should not
> "duplicate" MUST/SHALL/SHOULD/etc that are already in RFC3261.
> 
> We can say something: "The addressing of OPTIONS requests to a proxy is
> described in section X in RFC 3261"
> 
> ...or something like that.
> 
> Regards,
> 
> Christer
> 
> 
> 
> 
> > On May 5, 2010, at 1:20 AM, Christer Holmberg wrote:
> >
> > >
> > > Hi,
> > >
> > > As said earlier, 3261 already says that OPTIONS addressed
> > to proxies are to use sip:hostport, so I don't think we need
> > to say anything about that.
> > >
> > > Regards,
> > >
> > > Christer
> > >
> > >> -----Original Message-----
> > >> From: sipcore-bounces@ietf.org
> > >> [mailto:sipcore-bounces@ietf.org] On Behalf Of Paul E. Jones
> > >> Sent: 5. toukokuuta 2010 7:49
> > >> To: sipcore@ietf.org
> > >> Subject: [sipcore] draft-jones-sip-options-ping-01
> > >>
> > >> Folks,
> > >>
> > >> I posted a revised version of the OPTIONS "ping" draft
> > with changes
> > >> as per our discussions.  One change I made was that I
> > re-instated the
> > >> paragraph related to addressing, saying OPTIONS "should"
> > be addressed
> > >> to sip:hostport.
> > >> I'm still willing to consider removing that entirely if that's
> > >> preferred, but I recall that Hadriel preferred to have
> > something that
> > >> one could generally assume to be the proper address.  As a
> > means of
> > >> querying UAS status, I think that's a reasonable choice.
> > >>
> > >> Anyway, I welcome comments on the draft.
> > >>
> > >> Paul
> > >>
> > >>
> > >> _______________________________________________
> > >> 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 christer.holmberg@ericsson.com  Thu May  6 12:52:33 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: SIPCORE <sipcore@ietf.org>
Date: Thu, 6 May 2010 21:51:24 +0200
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Hi,

Any comments on this? It is going to be discussed at the 3GPP meeting next =
week, so some guidance from the WG would be useful, I think.

Regards,

Christer

________________________________________
From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On Behalf Of Chri=
ster Holmberg [christer.holmberg@ericsson.com]
Sent: Thursday, April 29, 2010 8:37 PM
To: SIPCORE
Subject: [sipcore] SigComp and Service-Route

Hi,

I appologize if the following issue have been discussed and solved before.

When reading RFC 5049 (Sigcomp), the UA figures out whether the outbound
proxy supports sigcomp by looking at outbound proxie's entry in the
Service-Route in the REGISTER response.

My questions are:

1. How can the UA ensure that the outbound proxy will be listed in the
Service-Route in the first place? RFC 3608 (Service-Route) does not
mandate the registrar to insert any intermediaries in the Service-Route,
e.g based on Path. It's all based on local policy.

2. Even if the registrar would build the Service-Route based on Path,
how does it know that the Path address/port values will be
usable/routable in the UA-towards-registrar direction (Path is supposed
to be used in the registrar-towards-UA direction)? This is similar to
the Record-Route problem, and intermediaries with different network
addresses/ports per direction would have to insert "double Path".

I guess the outbound proxy could insert itself in the Service-Route, but
that is not described anywhere in RFC 5049, and it is also a SHOULD NOT
in RFC 3608.

Regards,

Christer
_______________________________________________
sipcore mailing list
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> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Thursday, May 06, 2010 8:57 AM
> To: Hadriel Kaplan
>=20
> If you dig into it, the notion of "addressing OPTIONS to a proxy" is not
> a meaningful concept, so defining rules for how to do it is meaningless.

Agreed.
=20

> So fundamentally we need to just be talking about what it means to
> address a specific server, without regard to what sort of server it
> might be. And IMO that simply means using a URI with a domain that
> server "is responsible for". All URIs with that domain are in some sense
> addressed to that box, regardless of user part. At that point it is up
> to that box to interpret the user part as it sees fit. Some user parts
> may be treated by it as identifiers of certain resources within the box.
> Others might be treated as requiring retargeting. Note that even a URI
> with *no* user part could be treated as requiring retargeting. (E.g. the
> box might be a load balancer.)

I'm not exactly clear on what you're proposing. =20

If you're proposing the request be set to "sip:domain.com" or "sip:192.1.2.=
3", that hasn't worked in practice and probably can't without an indicator =
the OPTIONS is for status/ping. =20

If you're proposing to use "sip:special-user@domain.com" or "sip:special-us=
er@192.1.2.3", that's what I was originally proposing, except I was proposi=
ng "special-user" be a well-known reserved name.

But on thinking about it more I concurred with your original proposal sever=
al emails ago, to just have a new SIP Header to indicate the OPTIONS is for=
 the status/ping. (or we could make it a more generic header which says som=
ething like "this is for the system's administrative process", so we could =
use it in SUBSCRIBEs, for example)

-hadriel

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Cc: "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] draft-jones-sip-options-ping-01
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At end...

Hadriel Kaplan wrote:
> 
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>> Sent: Thursday, May 06, 2010 8:57 AM
>> To: Hadriel Kaplan
>>
>> If you dig into it, the notion of "addressing OPTIONS to a proxy" is not
>> a meaningful concept, so defining rules for how to do it is meaningless.
> 
> Agreed.
>  
> 
>> So fundamentally we need to just be talking about what it means to
>> address a specific server, without regard to what sort of server it
>> might be. And IMO that simply means using a URI with a domain that
>> server "is responsible for". All URIs with that domain are in some sense
>> addressed to that box, regardless of user part. At that point it is up
>> to that box to interpret the user part as it sees fit. Some user parts
>> may be treated by it as identifiers of certain resources within the box.
>> Others might be treated as requiring retargeting. Note that even a URI
>> with *no* user part could be treated as requiring retargeting. (E.g. the
>> box might be a load balancer.)
> 
> I'm not exactly clear on what you're proposing.  
> 
> If you're proposing the request be set to "sip:domain.com" or "sip:192.1.2.3", that hasn't worked in practice and probably can't without an indicator the OPTIONS is for status/ping.  
> 
> If you're proposing to use "sip:special-user@domain.com" or "sip:special-user@192.1.2.3", that's what I was originally proposing, except I was proposing "special-user" be a well-known reserved name.
> 
> But on thinking about it more I concurred with your original proposal several emails ago, to just have a new SIP Header to indicate the OPTIONS is for the status/ping. (or we could make it a more generic header which says something like "this is for the system's administrative process", so we could use it in SUBSCRIBEs, for example)

Sorry, I guess my comment left things dangling.

My point was that there currently is no consistent and well known way of 
identifying "the box". There isn't even a *notion* of addressing "the 
box", as distinct from any of the resources it manages.

One answer is that if you want to ping something, you need to find 
something to ping that has the properties you want, and that will be 
identified by *some* URI, but not the same one for every box, and maybe 
none for some boxes. So you need to find out by provisioning what it is 
that you should address your pings to.

I don't find that answer very satisfying.

Alternately, we have to come up with a unique way of addressing the ping 
to something with the appropriate properties, without appropriating a 
username value that some might object to.

I *guess* we could pick some obscure value for the user name for this 
purpose. But I hate appropriating special values out of someone else's 
namespace. Rather I think we should use some namespace that we control.

I see several possibilities:
- a new header indicating that this (OPTIONS) request has a
   special meaning

- a special URI parameter indicating that this request is addressed
   to a special function rather than to the normal resource
   associated with the URI

- a new request type (e.g. PING) in which case we get to specify
   all the semantics

I can find no particular benefit to the URI parameter over the header 
parameter, and it raises various other questions I'd just as soon avoid 
going into.

I have some fondness for a new method, but find the new header on 
OPTIONS to be roughly as good, and more people seem to like it, so I'm 
ok with that.

I do however appreciate the points that Cullen has raised, and would 
like to see them addressed before we go much further with this.

	Thanks,
	Paul

From mary.ietf.barnes@gmail.com  Fri May  7 13:22:22 2010
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Subject: [sipcore] 4244bis issue summary/proposed changes
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Hi all,

This is a follow-up to the discussion we had on 4244bis in Anaheim and
some subsequent discussion on the mailing list.  The primary point of
contention was around the support for voicemail.  The example we have
in the WG -00  describes the handling as based on policy and provides
two examples of such.  However, that's really not the best approach
and indeed voicemail really has two basic use cases that are specific
to configuration and thus a UAS for a specific configuration does have
determinant behavior and we can specify the exact mechanism by which
the appropriate information is derived from the History-Info header.

Our proposal is actually to move the majority of the use cases out of
this document into a separate informational (or BCP) document and add
more illustrative text and precise descriptions as to how the
necessary information  is derived from the History-Info header.  We
will add general text to 4244bis describing basic rules for using the
information in the UAS and application considerations sections.   We
will keep the basic examples from RFC 4244. In one sense, we are
reverting to what we had in earlier versions, except that the call
flows will be kept in sync with 4244bis.  We can decide where the call
flows should live once we get the updated documents submitted.

For background, here's a link to the earlier thread:
http://www.ietf.org/mail-archive/web/sipcore/current/msg02180.html

and meeting minutes:
http://www.ietf.org/proceedings/10mar/minutes/sipcore.html

We'll try to get the updates done within the next couple weeks.

Thanks,
Mary.

From pkyzivat@cisco.com  Fri May  7 14:32:00 2010
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Mary Barnes wrote:
> Hi all,
> 
> This is a follow-up to the discussion we had on 4244bis in Anaheim and
> some subsequent discussion on the mailing list.  The primary point of
> contention was around the support for voicemail.  The example we have
> in the WG -00  describes the handling as based on policy and provides
> two examples of such.  However, that's really not the best approach
> and indeed voicemail really has two basic use cases that are specific
> to configuration and thus a UAS for a specific configuration does have
> determinant behavior and we can specify the exact mechanism by which
> the appropriate information is derived from the History-Info header.

I look forward to hearing about that.

	Thanks,
	Paul

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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           : Re-INVITE and Target-refresh Request Handling in the Session Initiation Protocol (SIP)
	Author(s)       : G. Camarillo, et al.
	Filename        : draft-ietf-sipcore-reinvite-04.txt
	Pages           : 24
	Date            : 2010-05-08

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
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To: Paul Kyzivat <pkyzivat@cisco.com>, Cullen Jennings <fluffy@cisco.com>
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Subject: Re: [sipcore] draft-jones-sip-options-ping-01
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> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Friday, May 07, 2010 12:58 PM
>=20
> My point was that there currently is no consistent and well known way of
> identifying "the box". There isn't even a *notion* of addressing "the
> box", as distinct from any of the resources it manages.
> One answer is that if you want to ping something, you need to find
> something to ping that has the properties you want, and that will be
> identified by *some* URI, but not the same one for every box, and maybe
> none for some boxes. So you need to find out by provisioning what it is
> that you should address your pings to.
>=20
> I don't find that answer very satisfying.

Me neither.
=20

> I do however appreciate the points that Cullen has raised, and would
> like to see them addressed before we go much further with this.

Although I agree that the overload-chartered work is meant to improve thing=
s from the on/off flapping model we have today, I don't see how that invali=
dates the need or value of clearing up OPTIONS.  In some ways it's orthogon=
al, because the addressing problem still exists for OPTIONS regardless.

For sake of argument, ignore the "ping" aspect of OPTIONS.  Let's just say =
you wanted to send an OPTIONS to a SIP device (proxy, B2BUA, or UA) to lear=
n its capabilities - the *system's* capabilities, not a particular user ins=
tance in it.  How would you do that?  In practice today: through manual con=
figuration.  So let's say we went and created a draft to fix that, by using=
 a new header.  In that same draft, we re-iterated or clarified what the re=
sponses to such a query would mean.  In effect, that's Paul's draft.

Furthermore, if we cleared up how OPTIONS can be addressed, one could use a=
n OPTIONS to exchange the new overload info.  And likewise, a SUBSCRIBE for=
 overload info has the same issues OPTIONS does in terms of addressing.

-hadriel

From pkyzivat@cisco.com  Mon May 10 07:17:35 2010
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From: Paul Kyzivat <pkyzivat@cisco.com>
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To: Hadriel Kaplan <HKaplan@acmepacket.com>
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Cc: "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] draft-jones-sip-options-ping-01
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Inline...

Hadriel Kaplan wrote:
> 
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>> Sent: Friday, May 07, 2010 12:58 PM
>>
>> My point was that there currently is no consistent and well known way of
>> identifying "the box". There isn't even a *notion* of addressing "the
>> box", as distinct from any of the resources it manages.
>> One answer is that if you want to ping something, you need to find
>> something to ping that has the properties you want, and that will be
>> identified by *some* URI, but not the same one for every box, and maybe
>> none for some boxes. So you need to find out by provisioning what it is
>> that you should address your pings to.
>>
>> I don't find that answer very satisfying.
> 
> Me neither.
>  
> 
>> I do however appreciate the points that Cullen has raised, and would
>> like to see them addressed before we go much further with this.
> 
> Although I agree that the overload-chartered work is meant to improve things from the on/off flapping model we have today, I don't see how that invalidates the need or value of clearing up OPTIONS.  In some ways it's orthogonal, because the addressing problem still exists for OPTIONS regardless.
> 
> For sake of argument, ignore the "ping" aspect of OPTIONS.  Let's just say you wanted to send an OPTIONS to a SIP device (proxy, B2BUA, or UA) to learn its capabilities - the *system's* capabilities, not a particular user instance in it.  How would you do that?  In practice today: through manual configuration.  So let's say we went and created a draft to fix that, by using a new header.  In that same draft, we re-iterated or clarified what the responses to such a query would mean.  In effect, that's Paul's draft.

My counter-argument here is that the "system capabilities of a box" 
isn't necessarily a well defined concept. There may just be a 
multiplexer, based on the user part of the uri, to a variety of software 
agents. So no matter what you address the request to you are actually 
getting the status of one software agent or another, not of the box as a 
whole.

I interpret 3261 as saying the those elements "responsible for a domain" 
are the masters of the URIs that are valid in that domain. Others, that 
are not responsible for the domain, have no business "fabricating" a URI 
by pasting together a user part of their choice with the domain name. 
Instead they should be using addresses that are "given to them" directly 
or indirectly by an "agent" of the domain.

There is one case I know of where that is formally not the case. It is 
when using REGISTER, where a sip URI without user part is the R-URI. 
That is a restriction for domains that support REGISTER, and potentially 
may apply only to REGISTER.

Of course, this philosophy is violated all over the place. It can be 
justified in some cases as a matter that in some cases knowledge of the 
policy of a particular domain may extend to elements beyond those 
"responsible for the domain". But that still doesn't justify making such 
assumptions about a domain you have no special information about.

(And of course you an violate this is you are prepared for the 
consequences. If I didn't know Hadriel's sip URI, I might just try a 
variety of user parts in sip:*@acmepacket.com it hopes of guessing it.)

I think it is a big philosophical step to say that there is a reserved 
sip uri that will be serviced by *something* that can speak for the box 
as a whole.

> Furthermore, if we cleared up how OPTIONS can be addressed, one could use an OPTIONS to exchange the new overload info.  And likewise, a SUBSCRIBE for overload info has the same issues OPTIONS does in terms of addressing.

I wonder if that doesn't have some of the same issues - that a box may 
server as a multiplexer for a number of software agents. Much or all of 
the "overload" attributes may be attributable to those software agents, 
not to the box in general. I can certainly imagine architectures where 
requests to sip:X@example.com is overloaded, while requests to 
sip:Y@example.com is not overloaded.

But I haven't been party to the overload work, so I don't know what the 
thinking is like in that regard.

To me this seems like a reason to include the ping clarification (at 
least with respect "addressing the box") into the overload work. And to 
a certain extent that may also be true for the question of inconsistent 
responses to conditions, since the inconsistent responses of concern are 
all (I think) related to overload and up/down type conditions.

	Thanks,
	Paul

From radhika.r.roy@us.army.mil  Mon May 10 07:37:58 2010
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Cc: "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [sipcore] draft-jones-sip-options-ping-01
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Hi, Paul K.

Does it not mean that the very concept of Box is misleading?

In summary, does it not mean that we should rather keep our focus on "Agent" as a point of basic reference that controls Domain?

Best regards,
Radhika

----- Original Message -----
From: Paul Kyzivat <pkyzivat@cisco.com>
Date: Monday, May 10, 2010 10:17
Subject: Re: [sipcore] draft-jones-sip-options-ping-01
To: Hadriel Kaplan <HKaplan@acmepacket.com>
Cc: "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>


> Inline...
> 
> Hadriel Kaplan wrote:
> > 
> >> -----Original Message-----
> >> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> >> Sent: Friday, May 07, 2010 12:58 PM
> >>
> >> My point was that there currently is no consistent and well 
> known way of
> >> identifying "the box". There isn't even a *notion* of 
> addressing "the
> >> box", as distinct from any of the resources it manages.
> >> One answer is that if you want to ping something, you need to find
> >> something to ping that has the properties you want, and that 
> will be
> >> identified by *some* URI, but not the same one for every box, 
> and maybe
> >> none for some boxes. So you need to find out by provisioning 
> what it is
> >> that you should address your pings to.
> >>
> >> I don't find that answer very satisfying.
> > 
> > Me neither.
> > 
> > 
> >> I do however appreciate the points that Cullen has raised, and 
> would>> like to see them addressed before we go much further with 
> this.> 
> > Although I agree that the overload-chartered work is meant to 
> improve things from the on/off flapping model we have today, I 
> don't see how that invalidates the need or value of clearing up 
> OPTIONS. In some ways it's orthogonal, because the addressing 
> problem still exists for OPTIONS regardless.
> > 
> > For sake of argument, ignore the "ping" aspect of OPTIONS. 
> Let's just say you wanted to send an OPTIONS to a SIP device 
> (proxy, B2BUA, or UA) to learn its capabilities - the *system's* 
> capabilities, not a particular user instance in it. How would you 
> do that? In practice today: through manual configuration. So 
> let's say we went and created a draft to fix that, by using a new 
> header. In that same draft, we re-iterated or clarified what the 
> responses to such a query would mean. In effect, that's Paul's draft.
> 
> My counter-argument here is that the "system capabilities of a 
> box" 
> isn't necessarily a well defined concept. There may just be a 
> multiplexer, based on the user part of the uri, to a variety of 
> software 
> agents. So no matter what you address the request to you are 
> actually 
> getting the status of one software agent or another, not of the 
> box as a 
> whole.
> 
> I interpret 3261 as saying the those elements "responsible for a 
> domain" 
> are the masters of the URIs that are valid in that domain. Others, 
> that 
> are not responsible for the domain, have no business "fabricating" 
> a URI 
> by pasting together a user part of their choice with the domain 
> name. 
> Instead they should be using addresses that are "given to them" 
> directly 
> or indirectly by an "agent" of the domain.
> 
> There is one case I know of where that is formally not the case. 
> It is 
> when using REGISTER, where a sip URI without user part is the R-
> URI. 
> That is a restriction for domains that support REGISTER, and 
> potentially 
> may apply only to REGISTER.
> 
> Of course, this philosophy is violated all over the place. It can 
> be 
> justified in some cases as a matter that in some cases knowledge 
> of the 
> policy of a particular domain may extend to elements beyond those 
> "responsible for the domain". But that still doesn't justify 
> making such 
> assumptions about a domain you have no special information about.
> 
> (And of course you an violate this is you are prepared for the 
> consequences. If I didn't know Hadriel's sip URI, I might just try 
> a 
> variety of user parts in sip:*@acmepacket.com it hopes of guessing 
> it.)
> I think it is a big philosophical step to say that there is a 
> reserved 
> sip uri that will be serviced by *something* that can speak for 
> the box 
> as a whole.
> 
> > Furthermore, if we cleared up how OPTIONS can be addressed, one 
> could use an OPTIONS to exchange the new overload info. And 
> likewise, a SUBSCRIBE for overload info has the same issues 
> OPTIONS does in terms of addressing.
> 
> I wonder if that doesn't have some of the same issues - that a box 
> may 
> server as a multiplexer for a number of software agents. Much or 
> all of 
> the "overload" attributes may be attributable to those software 
> agents, 
> not to the box in general. I can certainly imagine architectures 
> where 
> requests to sip:X@example.com is overloaded, while requests to 
> sip:Y@example.com is not overloaded.
> 
> But I haven't been party to the overload work, so I don't know 
> what the 
> thinking is like in that regard.
> 
> To me this seems like a reason to include the ping clarification 
> (at 
> least with respect "addressing the box") into the overload work. 
> And to 
> a certain extent that may also be true for the question of 
> inconsistent 
> responses to conditions, since the inconsistent responses of 
> concern are 
> all (I think) related to overload and up/down type conditions.
> 
> Thanks,
> Paul
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore

From paulej@packetizer.com  Mon May 10 12:40:04 2010
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From: "Paul E. Jones" <paulej@packetizer.com>
To: "'Roy, Radhika R Dr CIV USA USAMC'" <radhika.r.roy@us.army.mil>, "'Paul Kyzivat'" <pkyzivat@cisco.com>
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Cc: sipcore@ietf.org, 'Hadriel Kaplan' <HKaplan@acmepacket.com>
Subject: Re: [sipcore] draft-jones-sip-options-ping-01
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Radhika, Paul,

I agree that the concept of a "box" is vague.  There is, though, an
addressable user agent to which we can send an OPTIONS message to query for
its operational status.  Within the subject draft, that's what we propose to
do.

Perhaps even using the word "propose" is not accurate, as this draft was
written in response to what clearly appears to be behavior already widely
implemented.  Unfortunately, it's not consistently implemented.  For
example, some devices return a 486, 503, or even 600 when they wish to
report that they're in an state where they cannot accept new dialogs.

I believe work on overload is important, but the purpose of overload is
rather different than sending an OPTIONS to a user agent to determine its
operational state.  The work on "keep" is also important, but the focus of
this draft is not to maintain open pinholes through a firewall.

The scope of the subject draft is purposely narrow.  Arguably, it's not
needed since RFC 3261 already defines these procedures.  Our assertion,
though, is that RFC 3261 does not go far enough in providing clarity so as
to ensure that the response to an OPTIONS "ping" is meaningful.  Perhaps
this draft doesn't, either, but recognize that this is just a draft and
we're prepared to make changes to improve clarity.

Paul

> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf Of Roy, Radhika R Dr CIV USA USAMC
> Sent: Monday, May 10, 2010 10:38 AM
> To: Paul Kyzivat
> Cc: Paul E. Jones; sipcore@ietf.org; Hadriel Kaplan
> Subject: Re: [sipcore] draft-jones-sip-options-ping-01
> 
> Hi, Paul K.
> 
> Does it not mean that the very concept of Box is misleading?
> 
> In summary, does it not mean that we should rather keep our focus on
> "Agent" as a point of basic reference that controls Domain?
> 
> Best regards,
> Radhika
> 
> ----- Original Message -----
> From: Paul Kyzivat <pkyzivat@cisco.com>
> Date: Monday, May 10, 2010 10:17
> Subject: Re: [sipcore] draft-jones-sip-options-ping-01
> To: Hadriel Kaplan <HKaplan@acmepacket.com>
> Cc: "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org"
> <sipcore@ietf.org>
> 
> 
> > Inline...
> >
> > Hadriel Kaplan wrote:
> > >
> > >> -----Original Message-----
> > >> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> > >> Sent: Friday, May 07, 2010 12:58 PM
> > >>
> > >> My point was that there currently is no consistent and well
> > known way of
> > >> identifying "the box". There isn't even a *notion* of
> > addressing "the
> > >> box", as distinct from any of the resources it manages.
> > >> One answer is that if you want to ping something, you need to find
> > >> something to ping that has the properties you want, and that
> > will be
> > >> identified by *some* URI, but not the same one for every box,
> > and maybe
> > >> none for some boxes. So you need to find out by provisioning
> > what it is
> > >> that you should address your pings to.
> > >>
> > >> I don't find that answer very satisfying.
> > >
> > > Me neither.
> > >
> > >
> > >> I do however appreciate the points that Cullen has raised, and
> > would>> like to see them addressed before we go much further with
> > this.>
> > > Although I agree that the overload-chartered work is meant to
> > improve things from the on/off flapping model we have today, I
> > don't see how that invalidates the need or value of clearing up
> > OPTIONS. In some ways it's orthogonal, because the addressing
> > problem still exists for OPTIONS regardless.
> > >
> > > For sake of argument, ignore the "ping" aspect of OPTIONS.
> > Let's just say you wanted to send an OPTIONS to a SIP device
> > (proxy, B2BUA, or UA) to learn its capabilities - the *system's*
> > capabilities, not a particular user instance in it. How would you
> > do that? In practice today: through manual configuration. So
> > let's say we went and created a draft to fix that, by using a new
> > header. In that same draft, we re-iterated or clarified what the
> > responses to such a query would mean. In effect, that's Paul's draft.
> >
> > My counter-argument here is that the "system capabilities of a
> > box"
> > isn't necessarily a well defined concept. There may just be a
> > multiplexer, based on the user part of the uri, to a variety of
> > software
> > agents. So no matter what you address the request to you are
> > actually
> > getting the status of one software agent or another, not of the
> > box as a
> > whole.
> >
> > I interpret 3261 as saying the those elements "responsible for a
> > domain"
> > are the masters of the URIs that are valid in that domain. Others,
> > that
> > are not responsible for the domain, have no business "fabricating"
> > a URI
> > by pasting together a user part of their choice with the domain
> > name.
> > Instead they should be using addresses that are "given to them"
> > directly
> > or indirectly by an "agent" of the domain.
> >
> > There is one case I know of where that is formally not the case.
> > It is
> > when using REGISTER, where a sip URI without user part is the R-
> > URI.
> > That is a restriction for domains that support REGISTER, and
> > potentially
> > may apply only to REGISTER.
> >
> > Of course, this philosophy is violated all over the place. It can
> > be
> > justified in some cases as a matter that in some cases knowledge
> > of the
> > policy of a particular domain may extend to elements beyond those
> > "responsible for the domain". But that still doesn't justify
> > making such
> > assumptions about a domain you have no special information about.
> >
> > (And of course you an violate this is you are prepared for the
> > consequences. If I didn't know Hadriel's sip URI, I might just try
> > a
> > variety of user parts in sip:*@acmepacket.com it hopes of guessing
> > it.)
> > I think it is a big philosophical step to say that there is a
> > reserved
> > sip uri that will be serviced by *something* that can speak for
> > the box
> > as a whole.
> >
> > > Furthermore, if we cleared up how OPTIONS can be addressed, one
> > could use an OPTIONS to exchange the new overload info. And
> > likewise, a SUBSCRIBE for overload info has the same issues
> > OPTIONS does in terms of addressing.
> >
> > I wonder if that doesn't have some of the same issues - that a box
> > may
> > server as a multiplexer for a number of software agents. Much or
> > all of
> > the "overload" attributes may be attributable to those software
> > agents,
> > not to the box in general. I can certainly imagine architectures
> > where
> > requests to sip:X@example.com is overloaded, while requests to
> > sip:Y@example.com is not overloaded.
> >
> > But I haven't been party to the overload work, so I don't know
> > what the
> > thinking is like in that regard.
> >
> > To me this seems like a reason to include the ping clarification
> > (at
> > least with respect "addressing the box") into the overload work.
> > And to
> > a certain extent that may also be true for the question of
> > inconsistent
> > responses to conditions, since the inconsistent responses of
> > concern are
> > all (I think) related to overload and up/down type conditions.
> >
> > Thanks,
> > Paul
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 



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Cc: sipcore@ietf.org, "'Roy, Radhika R Dr CIV USA USAMC'" <radhika.r.roy@us.army.mil>, 'Hadriel Kaplan' <HKaplan@acmepacket.com>
Subject: Re: [sipcore] draft-jones-sip-options-ping-01
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Paul E. Jones wrote:
> Radhika, Paul,
> 
> I agree that the concept of a "box" is vague.  There is, though, an
> addressable user agent to which we can send an OPTIONS message to query for
> its operational status.  Within the subject draft, that's what we propose to
> do.
> 
> Perhaps even using the word "propose" is not accurate, as this draft was
> written in response to what clearly appears to be behavior already widely
> implemented.  Unfortunately, it's not consistently implemented.  For
> example, some devices return a 486, 503, or even 600 when they wish to
> report that they're in an state where they cannot accept new dialogs.
> 
> I believe work on overload is important, but the purpose of overload is
> rather different than sending an OPTIONS to a user agent to determine its
> operational state. 

I'm still struggling to understand the difference.
With overload control the sender is trying to avoid sending to a box 
that is "overloaded". In the extreme, and overloaded box is pretty 
similar to a "down" box. We also want to avoid sending messages to a 
"down" box.

And overload status may come in degrees. So when given a choice, the 
sender may choose to send to the "least loaded" box.

So using OPTIONS to determine this sort of thing is pretty closely 
aligned with overload control. What are the other uses of OPTIONS ping 
that are *not* in this category?

> The work on "keep" is also important, but the focus of
> this draft is not to maintain open pinholes through a firewall.
> 
> The scope of the subject draft is purposely narrow.  Arguably, it's not
> needed since RFC 3261 already defines these procedures.  Our assertion,
> though, is that RFC 3261 does not go far enough in providing clarity so as
> to ensure that the response to an OPTIONS "ping" is meaningful.  Perhaps
> this draft doesn't, either, but recognize that this is just a draft and
> we're prepared to make changes to improve clarity.

At this point I think several people seem to have somewhat different 
conceptions of what is missing and what is needed. There may indeed be a 
need here. It just seems like so far we have a solution proposed while 
we as yet don't have clarity on what it is the solution *to*.

	Thanks,
	Paul

> Paul
> 
>> -----Original Message-----
>> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
>> Behalf Of Roy, Radhika R Dr CIV USA USAMC
>> Sent: Monday, May 10, 2010 10:38 AM
>> To: Paul Kyzivat
>> Cc: Paul E. Jones; sipcore@ietf.org; Hadriel Kaplan
>> Subject: Re: [sipcore] draft-jones-sip-options-ping-01
>>
>> Hi, Paul K.
>>
>> Does it not mean that the very concept of Box is misleading?
>>
>> In summary, does it not mean that we should rather keep our focus on
>> "Agent" as a point of basic reference that controls Domain?
>>
>> Best regards,
>> Radhika
>>
>> ----- Original Message -----
>> From: Paul Kyzivat <pkyzivat@cisco.com>
>> Date: Monday, May 10, 2010 10:17
>> Subject: Re: [sipcore] draft-jones-sip-options-ping-01
>> To: Hadriel Kaplan <HKaplan@acmepacket.com>
>> Cc: "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org"
>> <sipcore@ietf.org>
>>
>>
>>> Inline...
>>>
>>> Hadriel Kaplan wrote:
>>>>> -----Original Message-----
>>>>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>>>>> Sent: Friday, May 07, 2010 12:58 PM
>>>>>
>>>>> My point was that there currently is no consistent and well
>>> known way of
>>>>> identifying "the box". There isn't even a *notion* of
>>> addressing "the
>>>>> box", as distinct from any of the resources it manages.
>>>>> One answer is that if you want to ping something, you need to find
>>>>> something to ping that has the properties you want, and that
>>> will be
>>>>> identified by *some* URI, but not the same one for every box,
>>> and maybe
>>>>> none for some boxes. So you need to find out by provisioning
>>> what it is
>>>>> that you should address your pings to.
>>>>>
>>>>> I don't find that answer very satisfying.
>>>> Me neither.
>>>>
>>>>
>>>>> I do however appreciate the points that Cullen has raised, and
>>> would>> like to see them addressed before we go much further with
>>> this.>
>>>> Although I agree that the overload-chartered work is meant to
>>> improve things from the on/off flapping model we have today, I
>>> don't see how that invalidates the need or value of clearing up
>>> OPTIONS. In some ways it's orthogonal, because the addressing
>>> problem still exists for OPTIONS regardless.
>>>> For sake of argument, ignore the "ping" aspect of OPTIONS.
>>> Let's just say you wanted to send an OPTIONS to a SIP device
>>> (proxy, B2BUA, or UA) to learn its capabilities - the *system's*
>>> capabilities, not a particular user instance in it. How would you
>>> do that? In practice today: through manual configuration. So
>>> let's say we went and created a draft to fix that, by using a new
>>> header. In that same draft, we re-iterated or clarified what the
>>> responses to such a query would mean. In effect, that's Paul's draft.
>>>
>>> My counter-argument here is that the "system capabilities of a
>>> box"
>>> isn't necessarily a well defined concept. There may just be a
>>> multiplexer, based on the user part of the uri, to a variety of
>>> software
>>> agents. So no matter what you address the request to you are
>>> actually
>>> getting the status of one software agent or another, not of the
>>> box as a
>>> whole.
>>>
>>> I interpret 3261 as saying the those elements "responsible for a
>>> domain"
>>> are the masters of the URIs that are valid in that domain. Others,
>>> that
>>> are not responsible for the domain, have no business "fabricating"
>>> a URI
>>> by pasting together a user part of their choice with the domain
>>> name.
>>> Instead they should be using addresses that are "given to them"
>>> directly
>>> or indirectly by an "agent" of the domain.
>>>
>>> There is one case I know of where that is formally not the case.
>>> It is
>>> when using REGISTER, where a sip URI without user part is the R-
>>> URI.
>>> That is a restriction for domains that support REGISTER, and
>>> potentially
>>> may apply only to REGISTER.
>>>
>>> Of course, this philosophy is violated all over the place. It can
>>> be
>>> justified in some cases as a matter that in some cases knowledge
>>> of the
>>> policy of a particular domain may extend to elements beyond those
>>> "responsible for the domain". But that still doesn't justify
>>> making such
>>> assumptions about a domain you have no special information about.
>>>
>>> (And of course you an violate this is you are prepared for the
>>> consequences. If I didn't know Hadriel's sip URI, I might just try
>>> a
>>> variety of user parts in sip:*@acmepacket.com it hopes of guessing
>>> it.)
>>> I think it is a big philosophical step to say that there is a
>>> reserved
>>> sip uri that will be serviced by *something* that can speak for
>>> the box
>>> as a whole.
>>>
>>>> Furthermore, if we cleared up how OPTIONS can be addressed, one
>>> could use an OPTIONS to exchange the new overload info. And
>>> likewise, a SUBSCRIBE for overload info has the same issues
>>> OPTIONS does in terms of addressing.
>>>
>>> I wonder if that doesn't have some of the same issues - that a box
>>> may
>>> server as a multiplexer for a number of software agents. Much or
>>> all of
>>> the "overload" attributes may be attributable to those software
>>> agents,
>>> not to the box in general. I can certainly imagine architectures
>>> where
>>> requests to sip:X@example.com is overloaded, while requests to
>>> sip:Y@example.com is not overloaded.
>>>
>>> But I haven't been party to the overload work, so I don't know
>>> what the
>>> thinking is like in that regard.
>>>
>>> To me this seems like a reason to include the ping clarification
>>> (at
>>> least with respect "addressing the box") into the overload work.
>>> And to
>>> a certain extent that may also be true for the question of
>>> inconsistent
>>> responses to conditions, since the inconsistent responses of
>>> concern are
>>> all (I think) related to overload and up/down type conditions.
>>>
>>> Thanks,
>>> Paul
>>> _______________________________________________
>>> 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 pkyzivat@cisco.com  Mon May 10 14:10:42 2010
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Cc: "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [sipcore] draft-jones-sip-options-ping-01
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Roy, Radhika R Dr CIV USA USAMC wrote:
> Hi, Paul K.
> 
> Does it not mean that the very concept of Box is misleading?

There is nothing wrong with the "concept" of a box.
Its just that we have no sip artifacts that correlate to a box in a 
consistent way across implementations.

> In summary, does it not mean that we should rather keep our focus on "Agent" as a point of basic reference that controls Domain?

My use of "agent" was informal. We probably have similar problems 
defining it.

IIUC, the goal in all of this is to provide information via one request 
to some sip URI that is useful for predicting future behavior for some 
set of sip methods sent to some set of sip URIs.

The trick is identifying the mapping between the set of things you are 
interested in predicting the behavior of and the address you send the 
query to.

To get very far with this I think it will be necessary to restrict the 
architectures it applies to. Maybe there is some set of assumptions that 
are valid across a broad range of interesting implementations, or not. 
That may become apparent by trying to get agreement on what those 
assumptions are.

	Thanks,
	Paul

> Best regards,
> Radhika
> 
> ----- Original Message -----
> From: Paul Kyzivat <pkyzivat@cisco.com>
> Date: Monday, May 10, 2010 10:17
> Subject: Re: [sipcore] draft-jones-sip-options-ping-01
> To: Hadriel Kaplan <HKaplan@acmepacket.com>
> Cc: "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>
> 
> 
>> Inline...
>>
>> Hadriel Kaplan wrote:
>>>> -----Original Message-----
>>>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>>>> Sent: Friday, May 07, 2010 12:58 PM
>>>>
>>>> My point was that there currently is no consistent and well 
>> known way of
>>>> identifying "the box". There isn't even a *notion* of 
>> addressing "the
>>>> box", as distinct from any of the resources it manages.
>>>> One answer is that if you want to ping something, you need to find
>>>> something to ping that has the properties you want, and that 
>> will be
>>>> identified by *some* URI, but not the same one for every box, 
>> and maybe
>>>> none for some boxes. So you need to find out by provisioning 
>> what it is
>>>> that you should address your pings to.
>>>>
>>>> I don't find that answer very satisfying.
>>> Me neither.
>>>
>>>
>>>> I do however appreciate the points that Cullen has raised, and 
>> would>> like to see them addressed before we go much further with 
>> this.> 
>>> Although I agree that the overload-chartered work is meant to 
>> improve things from the on/off flapping model we have today, I 
>> don't see how that invalidates the need or value of clearing up 
>> OPTIONS. In some ways it's orthogonal, because the addressing 
>> problem still exists for OPTIONS regardless.
>>> For sake of argument, ignore the "ping" aspect of OPTIONS. 
>> Let's just say you wanted to send an OPTIONS to a SIP device 
>> (proxy, B2BUA, or UA) to learn its capabilities - the *system's* 
>> capabilities, not a particular user instance in it. How would you 
>> do that? In practice today: through manual configuration. So 
>> let's say we went and created a draft to fix that, by using a new 
>> header. In that same draft, we re-iterated or clarified what the 
>> responses to such a query would mean. In effect, that's Paul's draft.
>>
>> My counter-argument here is that the "system capabilities of a 
>> box" 
>> isn't necessarily a well defined concept. There may just be a 
>> multiplexer, based on the user part of the uri, to a variety of 
>> software 
>> agents. So no matter what you address the request to you are 
>> actually 
>> getting the status of one software agent or another, not of the 
>> box as a 
>> whole.
>>
>> I interpret 3261 as saying the those elements "responsible for a 
>> domain" 
>> are the masters of the URIs that are valid in that domain. Others, 
>> that 
>> are not responsible for the domain, have no business "fabricating" 
>> a URI 
>> by pasting together a user part of their choice with the domain 
>> name. 
>> Instead they should be using addresses that are "given to them" 
>> directly 
>> or indirectly by an "agent" of the domain.
>>
>> There is one case I know of where that is formally not the case. 
>> It is 
>> when using REGISTER, where a sip URI without user part is the R-
>> URI. 
>> That is a restriction for domains that support REGISTER, and 
>> potentially 
>> may apply only to REGISTER.
>>
>> Of course, this philosophy is violated all over the place. It can 
>> be 
>> justified in some cases as a matter that in some cases knowledge 
>> of the 
>> policy of a particular domain may extend to elements beyond those 
>> "responsible for the domain". But that still doesn't justify 
>> making such 
>> assumptions about a domain you have no special information about.
>>
>> (And of course you an violate this is you are prepared for the 
>> consequences. If I didn't know Hadriel's sip URI, I might just try 
>> a 
>> variety of user parts in sip:*@acmepacket.com it hopes of guessing 
>> it.)
>> I think it is a big philosophical step to say that there is a 
>> reserved 
>> sip uri that will be serviced by *something* that can speak for 
>> the box 
>> as a whole.
>>
>>> Furthermore, if we cleared up how OPTIONS can be addressed, one 
>> could use an OPTIONS to exchange the new overload info. And 
>> likewise, a SUBSCRIBE for overload info has the same issues 
>> OPTIONS does in terms of addressing.
>>
>> I wonder if that doesn't have some of the same issues - that a box 
>> may 
>> server as a multiplexer for a number of software agents. Much or 
>> all of 
>> the "overload" attributes may be attributable to those software 
>> agents, 
>> not to the box in general. I can certainly imagine architectures 
>> where 
>> requests to sip:X@example.com is overloaded, while requests to 
>> sip:Y@example.com is not overloaded.
>>
>> But I haven't been party to the overload work, so I don't know 
>> what the 
>> thinking is like in that regard.
>>
>> To me this seems like a reason to include the ping clarification 
>> (at 
>> least with respect "addressing the box") into the overload work. 
>> And to 
>> a certain extent that may also be true for the question of 
>> inconsistent 
>> responses to conditions, since the inconsistent responses of 
>> concern are 
>> all (I think) related to overload and up/down type conditions.
>>
>> Thanks,
>> Paul
>> _______________________________________________
>> sipcore mailing list
>> sipcore@ietf.org
>> https://www.ietf.org/mailman/listinfo/sipcore
> 

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Cc: sipcore@ietf.org, "'Roy, Radhika R Dr CIV USA USAMC'" <radhika.r.roy@us.army.mil>, 'Hadriel Kaplan' <HKaplan@acmepacket.com>
Subject: Re: [sipcore] draft-jones-sip-options-ping-01
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Paul,

> > I believe work on overload is important, but the purpose of overload
> is
> > rather different than sending an OPTIONS to a user agent to determine
> its
> > operational state.
> 
> I'm still struggling to understand the difference.
> With overload control the sender is trying to avoid sending to a box
> that is "overloaded". In the extreme, and overloaded box is pretty
> similar to a "down" box. We also want to avoid sending messages to a
> "down" box.

The OPTIONS method is not used only to determine is a user agent is
overloaded.  This is just one example we've discussed.

We're not proposing to make fine-grained routing decisions based over the
results of an OPTIONS response, though it certainly has some utility for
helping to facilitate traffic.  Not all devices need the same level of
granularity and responsiveness as what might be provided via overload.  Even
so, a service provider would likely appreciate knowing a box is currently in
a maintenance state so as to not waste time directing traffic to a
known-down device.

A 503 could tell the administrator that there is an issue and that code
would also be used when an administrator takes a box down for maintenance
(as per 3261).  Most certainly, a 500 likely would mean there is a problem
that needs immediate attention.  So, it has management value.

The OPTIONS "ping" could be used to check the status of an IP phone or ATA
device, neither of which are likely to implement the overload spec.
 
A service provider may not be very interested to know whether a small
business SBC is overloaded or not, but it might want to know whether it's
responsive in the least and what status it is reporting presently.  Even in
cases where two peer SBCs might want to know overload information, that does
not eliminate the desire for an administrative device to also want to
collect status information.  Managed service providers would likely want to
keep close tabs on the current state of a PBX or SBC, but that information
may or may not be collected by an SBC.

> And overload status may come in degrees. So when given a choice, the
> sender may choose to send to the "least loaded" box.

Yeah, and that's not what we're trying to address.
 
> So using OPTIONS to determine this sort of thing is pretty closely
> aligned with overload control. What are the other uses of OPTIONS ping
> that are *not* in this category?

See above.
 
> > The work on "keep" is also important, but the focus of
> > this draft is not to maintain open pinholes through a firewall.
> >
> > The scope of the subject draft is purposely narrow.  Arguably, it's
> not
> > needed since RFC 3261 already defines these procedures.  Our
> assertion,
> > though, is that RFC 3261 does not go far enough in providing clarity
> so as
> > to ensure that the response to an OPTIONS "ping" is meaningful.
> Perhaps
> > this draft doesn't, either, but recognize that this is just a draft
> and
> > we're prepared to make changes to improve clarity.
> 
> At this point I think several people seem to have somewhat different
> conceptions of what is missing and what is needed. There may indeed be
> a
> need here. It just seems like so far we have a solution proposed while
> we as yet don't have clarity on what it is the solution *to*.

For a variety of reasons (as mentioned above), there is a desire to learn
the operational status of a given user agent.  The method through which one
accomplishes this task is by sending an OPTIONS message as per RFC 3261.
This will report whether the device is operational (200) or overloaded or
down for maintenance (503).  In practice, though, we see lots of response
codes returned, just for these two basic states.

Since there are so many varied implementations all over the map, a desire
was expressed to be able to differentiate an OPTIONS "ping" from an OPTIONS
intended to query device capabilities.  Personally, I would make no
distinction in my code, but the fact is that we're dealing with an aging
protocol with a lot of legacy implementations we need to consider.
Therefore, I conceded on the request to make a distinction by including a
header (of some sort) to indicate that the message is a "ping".  For those
who do wish to implement two code paths in order to support different
behavior, they could use that header to do that.  On that point, I don't
really care so much: I'd encourage a single behavior, but I'm not going to
argue with folks who clearly see value in having something to use as a
differentiator: if they see a need, then one likely exists.

The bottom line is that we're not trying to define anything not already
defined in RFC 3261.  What we are trying to do is improve upon that text by
recommending response codes and clarifying the fact that OPTIONS may be used
for this purpose.  If we do not move forward with this draft, then we just
live with the varied implementations.  I do not think the work on overload
will likely eliminate use of the OPTIONS "ping" for the aforementioned use
cases, so we'll continue to see varying behavior.

Paul



From dean.willis@softarmor.com  Mon May 10 14:59:09 2010
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Subject: Re: [sipcore] FW: I-D Action:draft-jones-sip-options-ping-00.txt
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Paul E. Jones wrote:
> Dean,
> 
>> So for example, a new header field "Options-mode" with a value of
>> "ping" means "If you support this extension, do not proxy this
>> request, just send an immediate 200 OK response  and do not bother
>> populating the Supported header field of the 200 OK response."
>> Somebody who didn't recognize the header field would just send a
>> conventional 200 OK response.
> 
> Not objecting to the header, but I think we would have failed in our effort
> if the message might possibly be proxied.  The intent was for this to be
> peer-to-peer.

Then a header, combined with an options-tag used in a Prcy-Require"
would do the job.

>  
>> And perhaps an option tag of "optionsmodeping". That way we can add
>> further options-modes, and option tags for them in the future as (and
>> if) needed.
> 
> So, you want a head like this:
> 
>   Options-Mode: ping
> 
> Or this:
> 
>   Require: optionsmodeping

Both. And a Proxy-Require: optionmsmodeping to make sure that proxies
that understand Options-Mode: ping don't forward.

--
Dean

From dean.willis@softarmor.com  Mon May 10 15:19:46 2010
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Subject: Re: [sipcore] draft-jones-sip-options-ping-01
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Paul E. Jones wrote:
> Radhika, Paul,
> 
> I agree that the concept of a "box" is vague.  There is, though, an
> addressable user agent to which we can send an OPTIONS message to query for
> its operational status.  Within the subject draft, that's what we propose to
> do.
> 

The problem is that sip"alice@192.168.3.1", "sip:bob@192.168.3.1",
"sip:charlie@192.168.3.1", and "sip:ping@192.168.3.1" might actually be
four different SIP user agent instances, using four different software
stacks, all being demuxed by an independent proxy module running on port
5060 of "192.168.3.1", which might also be a UA for
"sip:admin@192.168.3.1" (or something entirely different), meaning that
it too isn't "just a proxy".

Which piece of software did you want to have answer the "ping" and how
is this not somehow different real behavior from what RFC 3261 says?

We can certainly alter the behavior of OPTIONS using option-tags and
headers. But this doesn't make it "automatically work" when the
recipient doesn't understand the extension. Rather, it makes the request
automatically (and detectably) fail, so that the UAC sending the OPTIONS
request can fall back to sending an old-fashioned OPTIONS if it wants to.

--
Dean

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Subject: Re: [sipcore] draft-jones-sip-options-ping-01
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Paul E. Jones wrote:
> Paul,
> 
>>> I believe work on overload is important, but the purpose of overload
>> is
>>> rather different than sending an OPTIONS to a user agent to determine
>> its
>>> operational state.
>> I'm still struggling to understand the difference.
>> With overload control the sender is trying to avoid sending to a box
>> that is "overloaded". In the extreme, and overloaded box is pretty
>> similar to a "down" box. We also want to avoid sending messages to a
>> "down" box.
> 
> The OPTIONS method is not used only to determine is a user agent is
> overloaded.  This is just one example we've discussed.

I think you are viewing "overload" more narrowly than I am.

> We're not proposing to make fine-grained routing decisions based over the
> results of an OPTIONS response, though it certainly has some utility for
> helping to facilitate traffic.  Not all devices need the same level of
> granularity and responsiveness as what might be provided via overload.  Even
> so, a service provider would likely appreciate knowing a box is currently in
> a maintenance state so as to not waste time directing traffic to a
> known-down device.

A box in maintenance state is effectively "down" if it won't accept 
additional calls. Though this may be subtle if it will still accept some 
class of requests, such as those that are in-dialog, while rejecting others.

But "wasting time" directing traffic to known-down devices is, AFAIK, a 
valid subject for overload control, since it loads the network and the 
sending device.

Maybe it is not part of the currently planned work on overload control.
But if not, then I would argue that we should consider broadening the 
charter of that WG, rather than inventing an entirely separate solution 
to another aspect of the same problem space.

> A 503 could tell the administrator that there is an issue and that code
> would also be used when an administrator takes a box down for maintenance
> (as per 3261).  Most certainly, a 500 likely would mean there is a problem
> that needs immediate attention.  So, it has management value.

A 500 has essentially no predictive value about how other requests will 
be processed in the future.

> The OPTIONS "ping" could be used to check the status of an IP phone or ATA
> device, neither of which are likely to implement the overload spec.

What would be the purpose of obtaining this information?
This is one of the primary uses for OPTIONS as it is defined in 3261. 
Its is expected to return the same result as an INVITE would. There are 
many reasons why that doesn't mean as much as it might seem, but I don't 
think we need a lot of clarification for that usage.

It might be more complicated if you don't send the OPTIONS to the same 
address as you would send an INVITE. That is because there is probably 
no particularly good answer in that case - at least without being much 
more precise about the intended use of the result.

Even dumb devices probably should implement some limited form of 
overload control, once that is defined. At least enough to protect the 
network.

> A service provider may not be very interested to know whether a small
> business SBC is overloaded or not,

It *ought* to be, of it isn't providing very good service.
And ignoring this and as a result reaching congestion collapse isn't 
good for it even if it doesn't care about the PBX.

> but it might want to know whether it's
> responsive in the least and what status it is reporting presently. 

Why would it want to know one and not the other?
If it doesn't care about overload control, wouldn't most likely not care 
whether the device is up either? If its not doing overload control, then 
the most interesting statistic is likely to be the number of calls 
delivered successfully, vs. the number that failed due to lack of 
response by the pbx.

Sure, it might want this information just because it is "curious". But 
we are going to have a hard time refining the definitions of these 
various "states" without some notion of how they are to be used.

> Even in
> cases where two peer SBCs might want to know overload information, that does
> not eliminate the desire for an administrative device to also want to
> collect status information.  Managed service providers would likely want to
> keep close tabs on the current state of a PBX or SBC, but that information
> may or may not be collected by an SBC.

All of the above cases you are mentioning are of interest *before* 
nailing down the mechanism to address them.

>> And overload status may come in degrees. So when given a choice, the
>> sender may choose to send to the "least loaded" box.
> 
> Yeah, and that's not what we're trying to address.
>  
>> So using OPTIONS to determine this sort of thing is pretty closely
>> aligned with overload control. What are the other uses of OPTIONS ping
>> that are *not* in this category?
> 
> See above.
>  
>>> The work on "keep" is also important, but the focus of
>>> this draft is not to maintain open pinholes through a firewall.
>>>
>>> The scope of the subject draft is purposely narrow.  Arguably, it's
>> not
>>> needed since RFC 3261 already defines these procedures.  Our
>> assertion,
>>> though, is that RFC 3261 does not go far enough in providing clarity
>> so as
>>> to ensure that the response to an OPTIONS "ping" is meaningful.
>> Perhaps
>>> this draft doesn't, either, but recognize that this is just a draft
>> and
>>> we're prepared to make changes to improve clarity.
>> At this point I think several people seem to have somewhat different
>> conceptions of what is missing and what is needed. There may indeed be
>> a
>> need here. It just seems like so far we have a solution proposed while
>> we as yet don't have clarity on what it is the solution *to*.
> 
> For a variety of reasons (as mentioned above), there is a desire to learn
> the operational status of a given user agent. 

I do believe there are a number of good reasons to want to know 
"operational status". I think most of them have to do with wanting to 
predict certain aspects of future behavior. But the devil is in the 
details: what aspects of future behavior do you wish to predict? What 
information would be useful for making that prediction? And who has that 
information?

The response code to an OPTIONS request will have some limited 
predictive value. Will refining the way options responses are made 
improve that predictive value?

Will it provide predictive value that is needed and not already 
available? Maybe, but I don't really know. Only if we know what kinds of 
predictions are to be made will we be able to determine that. And that 
is exactly the info that I am not seeing so far.

	Thanks,
	Paul K

> The method through which one
> accomplishes this task is by sending an OPTIONS message as per RFC 3261.
> This will report whether the device is operational (200) or overloaded or
> down for maintenance (503).  In practice, though, we see lots of response
> codes returned, just for these two basic states.
> 
> Since there are so many varied implementations all over the map, a desire
> was expressed to be able to differentiate an OPTIONS "ping" from an OPTIONS
> intended to query device capabilities.  Personally, I would make no
> distinction in my code, but the fact is that we're dealing with an aging
> protocol with a lot of legacy implementations we need to consider.
> Therefore, I conceded on the request to make a distinction by including a
> header (of some sort) to indicate that the message is a "ping".  For those
> who do wish to implement two code paths in order to support different
> behavior, they could use that header to do that.  On that point, I don't
> really care so much: I'd encourage a single behavior, but I'm not going to
> argue with folks who clearly see value in having something to use as a
> differentiator: if they see a need, then one likely exists.
> 
> The bottom line is that we're not trying to define anything not already
> defined in RFC 3261.  What we are trying to do is improve upon that text by
> recommending response codes and clarifying the fact that OPTIONS may be used
> for this purpose.  If we do not move forward with this draft, then we just
> live with the varied implementations.  I do not think the work on overload
> will likely eliminate use of the OPTIONS "ping" for the aforementioned use
> cases, so we'll continue to see varying behavior.
> 
> Paul
> 
> 
> 

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Subject: Re: [sipcore] draft-jones-sip-options-ping-01
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Dean,

> > I agree that the concept of a "box" is vague.  There is, though, an
> > addressable user agent to which we can send an OPTIONS message to
> query for
> > its operational status.  Within the subject draft, that's what we
> propose to
> > do.
> >
> 
> The problem is that sip"alice@192.168.3.1", "sip:bob@192.168.3.1",
> "sip:charlie@192.168.3.1", and "sip:ping@192.168.3.1" might actually be
> four different SIP user agent instances, using four different software
> stacks, all being demuxed by an independent proxy module running on
> port
> 5060 of "192.168.3.1", which might also be a UA for
> "sip:admin@192.168.3.1" (or something entirely different), meaning that
> it too isn't "just a proxy".

What I proposed was sip:hostport.  I understand that there may be a wide
variation of things done in practice, but one ought to be able to expect
some predictable behavior when sending an OPTIONS to 192.168.1.10 addressed
to sip:192.168.1.10.
 
> Which piece of software did you want to have answer the "ping" and how
> is this not somehow different real behavior from what RFC 3261 says?

Whatever software should respond to sip:192.168.1.10.  I'm not proposing any
behavior that is different than RFC 3261, though I am recommending the
preferred form of addressing and response value.
 
> We can certainly alter the behavior of OPTIONS using option-tags and
> headers. But this doesn't make it "automatically work" when the
> recipient doesn't understand the extension. Rather, it makes the
> request
> automatically (and detectably) fail, so that the UAC sending the
> OPTIONS
> request can fall back to sending an old-fashioned OPTIONS if it wants
> to.

I never proposed a different behavior, though some did.  In that case, we
could insert headers as you proposed, perhaps a header in the request and
response.  There were valid reasons for doing that with which I'll not
argue.

I recognize that it would take time be implemented widely, but I also don't
see that as an issue.  Remember, I'm proposing the use of this "ping"
between two devices that have some general knowledge of each other.  I would
not expect this to be sent out over the Internet.  Rather, it would be used
between a management system and devices it manages or between two SBCs, etc.
So, the scope is relatively narrow as compared to the interoperability
requirements of sending an INVITE out over the Internet.

Paul



From HKaplan@acmepacket.com  Mon May 10 21:14:33 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Dean Willis <dean.willis@softarmor.com>, "Paul E. Jones" <paulej@packetizer.com>
Date: Tue, 11 May 2010 00:14:13 -0400
Thread-Topic: [sipcore] draft-jones-sip-options-ping-01
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>, "'Roy, Radhika R Dr CIV USA USAMC'" <radhika.r.roy@us.army.mil>
Subject: Re: [sipcore] draft-jones-sip-options-ping-01
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> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: Monday, May 10, 2010 6:19 PM
> To: Paul E. Jones
> Cc: 'Roy, Radhika R Dr CIV USA USAMC'; 'Paul Kyzivat'; sipcore@ietf.org;
>=20
> Which piece of software did you want to have answer the "ping" and how
> is this not somehow different real behavior from what RFC 3261 says?

Right now, Paul is only stipulating what 3261 says, because he's proposing =
using "sip:hostport" and wants to query the logical "proxy" role of the nex=
t-hop.

RFC 3261 section 11 says:
   The target of the OPTIONS request is identified by the Request-URI,
   which could identify another UA or a SIP server.  If the OPTIONS is
   addressed to a proxy server, the Request-URI is set without a user
   part, similar to the way a Request-URI is set for a REGISTER request.

=20
> We can certainly alter the behavior of OPTIONS using option-tags and
> headers. But this doesn't make it "automatically work" when the
> recipient doesn't understand the extension. Rather, it makes the request
> automatically (and detectably) fail, so that the UAC sending the OPTIONS
> request can fall back to sending an old-fashioned OPTIONS if it wants to.

Paul's current draft proposes that we formulate the OPTIONS as per RFC 3261=
, such that it would be interpreted as "an old-fashioned" OPTIONS.

Some of us (maybe only me) feel there should be something different about t=
his particular OPTIONS, because we'd like to know the OPTIONS we received, =
or the response to one we sent, is actually following the draft's clarifica=
tions for response codes.  It doesn't *have* to do that - the whole thing c=
ould be based purely on configuration; but it would be *nice* to make this =
thing automatic by the existence of a header field.

-hadriel

From HKaplan@acmepacket.com  Mon May 10 21:55:21 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Paul Kyzivat <pkyzivat@cisco.com>, Cullen Jennings <fluffy@cisco.com>
Date: Tue, 11 May 2010 00:55:06 -0400
Thread-Topic: [sipcore] draft-jones-sip-options-ping-01
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Cc: "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] draft-jones-sip-options-ping-01
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OPTIONS already exists.  It is widely deployed.  It is widely deployed as a=
 way to query system status. (Arguably more so for that than for query of a=
ctual extensions/sdp/etc.)  RFC 3261 specifically described this use-case f=
or OPTIONS.  In practice, though, it has interop issues.  The interop issue=
s mostly revolve around response codes and target addressing.  Today, those=
 interop issues are resolved by manual configuration.  Paul is trying to al=
leviate those interop issues by recommending a particular configuration, wh=
ich follows rfc3261's language.  One could almost call it a BCP. (except th=
e term "BCP" means something different in IETF than normal nomenclature)

So he submitted a draft to do that.  The response he gets is "we're working=
 on an overload control mechanism and this thing has a bit of that... so no=
, we can't clean this up".

It's like you guys are saying "Oh your arm hurts when you lift stuff?  Well=
 we could fix that, but we're working on giving you a new hand, and that ho=
lds the things you lift, so stop using your arm while we go design this new=
 hand for it... oh and make sure you tell the hand designers about this arm=
 problem...".   No?

-hadriel

From dean.willis@softarmor.com  Mon May 10 22:08:44 2010
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Cc: "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>, "'Roy, Radhika R Dr CIV USA USAMC'" <radhika.r.roy@us.army.mil>
Subject: Re: [sipcore] draft-jones-sip-options-ping-01
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Hadriel Kaplan wrote:

>> We can certainly alter the behavior of OPTIONS using option-tags
>> and headers. But this doesn't make it "automatically work" when the
>>  recipient doesn't understand the extension. Rather, it makes the
>> request automatically (and detectably) fail, so that the UAC
>> sending the OPTIONS request can fall back to sending an
>> old-fashioned OPTIONS if it wants to.
> 
> Paul's current draft proposes that we formulate the OPTIONS as per
> RFC 3261, such that it would be interpreted as "an old-fashioned"
> OPTIONS.

If it's also being responded too as an old-fashioned OPTIONS and is
consistent with the content-rules of an old-fashioned OPTIONS and is
subject to the proxy-rules of an old-fashioned OPTIONS and is in fact,
in no way distinguishable from an old-fashioned OPTIONS, then you're
right and no extension is needed.


> Some of us (maybe only me) feel there should be something different
> about this particular OPTIONS, because we'd like to know the OPTIONS
> we received, or the response to one we sent, is actually following
> the draft's clarifications for response codes.  It doesn't *have* to
> do that - the whole thing could be based purely on configuration; but
> it would be *nice* to make this thing automatic by the existence of a
> header field.


Perhaps something that clarified the use cases for OPTIONS in general,
including targeting the request to a given node, preventing proxying of
the request using max-forwards, and clarifying the response codes for
different situations would be useful. Why just document the ping case?

And I'm still fuzzy on why you need to be able to differentiate a "ping
OPTIONS" from a "regular" OPTIONS. The use cases for doing this that
I've heard so far seem to require an extension.

--
Dean

From pkyzivat@cisco.com  Tue May 11 06:42:29 2010
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Cc: "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] draft-jones-sip-options-ping-01
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Hadriel Kaplan wrote:
> OPTIONS already exists.  It is widely deployed.  It is widely deployed as a way to query system status. (Arguably more so for that than for query of actual extensions/sdp/etc.)  RFC 3261 specifically described this use-case for OPTIONS.  In practice, though, it has interop issues.  The interop issues mostly revolve around response codes and target addressing.  Today, those interop issues are resolved by manual configuration.  Paul is trying to alleviate those interop issues by recommending a particular configuration, which follows rfc3261's language.  One could almost call it a BCP. (except the term "BCP" means something different in IETF than normal nomenclature)
> 
> So he submitted a draft to do that.  The response he gets is "we're working on an overload control mechanism and this thing has a bit of that... so no, we can't clean this up".
> 
> It's like you guys are saying "Oh your arm hurts when you lift stuff?  Well we could fix that, but we're working on giving you a new hand, and that holds the things you lift, so stop using your arm while we go design this new hand for it... oh and make sure you tell the hand designers about this arm problem...".   No?

Well, I guess the problem is that I am not getting answers to questions 
about *why* the options is being used - what one intends to accomplish 
with the answer. All I'm getting is "the answers that I get are 
inconsistent for my purposes". And so I am making my own assumptions 
about why, that are conflated with the overload work.

The fact is, there are many possible reasons to use OPTIONS. I suspect 
that the reason there are different responses to similar situations is 
at least in part because the person deciding what response to give made 
the decision based on his/her notion of what the response would be used for.

As I said yesterday, ISTM that one of the primary reasons for sending 
OPTIONS is so that the response to it can be used to make predictions 
about how future requests sent to related addresses will fare.

I could say more, but in the interest of reaching some level of common 
understanding, I won't. Instead, I'll just ask: is that the primary goal 
here?

	Thanks,
	Paul

From brett@broadsoft.com  Tue May 11 07:31:22 2010
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From: Brett Tate <brett@broadsoft.com>
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Date: Tue, 11 May 2010 07:29:05 -0700
Thread-Topic: rfc3428 section 10: request-uri does not reflect binding
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Subject: [sipcore] rfc3428 section 10: request-uri does not reflect binding
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Howdy,

Within RFC3428 section 10 example, the proxy "finds a binding from sip:user=
2@domain.com to sip:user2@user2pc.domain.com"; however it indicates sip:use=
r2@domain.com instead of sip:user2@user2pc.domain.com within proxied MESSAG=
E's request-uri.

I assume that this a bug within the example (similar to tag and extraneous =
';' issues); however I thought that I'd ask before opening errata... is suc=
h proxy behavior intentional?  If intentional, what is the corresponding no=
rmative text within RFC3428?

Thanks,
Brett


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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
Date: Tue, 11 May 2010 11:34:08 -0400
Thread-Topic: [sipcore] draft-jones-sip-options-ping-01
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Cc: "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] draft-jones-sip-options-ping-01
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> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Tuesday, May 11, 2010 9:41 AM
> To: Hadriel Kaplan
>=20
> As I said yesterday, ISTM that one of the primary reasons for sending
> OPTIONS is so that the response to it can be used to make predictions
> about how future requests sent to related addresses will fare.
> I could say more, but in the interest of reaching some level of common
> understanding, I won't. Instead, I'll just ask: is that the primary goal
> here?

For me it is.  I do also have a need to actually learn what the next-hop's =
supported extensions are (and I do mean "box-level" ones), but that's not t=
he main driver use-case for me at this instant in time. =20

So for me, the main use-case for sending OPTIONS right now is to learn the =
capability of the next-hop to process messages.  Load-related state, admin =
state, and actual transport layer reachability as well.  That's what many/m=
ost people have been doing with OPTIONS for the past many years, after all.=
  Paul J's not proposing to change that - just to clarify what 3261 says ab=
out it, so we don't have to do as much configuration to continue doing it.

-hadriel

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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Dean Willis <dean.willis@softarmor.com>
Date: Tue, 11 May 2010 11:53:26 -0400
Thread-Topic: [sipcore] draft-jones-sip-options-ping-01
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Subject: Re: [sipcore] draft-jones-sip-options-ping-01
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> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: Tuesday, May 11, 2010 1:08 AM
> To: Hadriel Kaplan
=20
> And I'm still fuzzy on why you need to be able to differentiate a "ping
> OPTIONS" from a "regular" OPTIONS. The use cases for doing this that
> I've heard so far seem to require an extension.

Basically because different vendors use different response codes and proces=
s the request slightly differently, but all are legal per 3261.  For exampl=
e, let's say vendor X sends this OPTIONS per 3261, and expects a 503 to mea=
n admin-down.  But some vendor Y responds to it with a 500 to mean admin-do=
wn.

Vendor Y is not going to change its default behavior, because they're deplo=
yed doing it that way and it's working.  So they need something in the requ=
est to tell them "this request is per draft foo, so use the specific respon=
se codes per draft foo".  Likewise, vendor X needs to know the responder di=
d draft foo, so it doesn't apply its legacy default behavior (whatever that=
 is) but follows draft foo instead.

So we could put some new param in the request's Via (like ";foo"), and have=
 the responder set it to something (like ";foo=3Dyes").  Or we could do it =
with a Header.  Either way, if the responder does not do foo both sides wil=
l just do their legacy default logic.

-hadriel

From pkyzivat@cisco.com  Tue May 11 10:51:20 2010
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Cc: "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: [sipcore] The predictive value of OPTIONS
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[Changing the subject line]

Hadriel Kaplan wrote:
> 
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
>> Sent: Tuesday, May 11, 2010 9:41 AM
>> To: Hadriel Kaplan
>>
>> As I said yesterday, ISTM that one of the primary reasons for sending
>> OPTIONS is so that the response to it can be used to make predictions
>> about how future requests sent to related addresses will fare.
>> I could say more, but in the interest of reaching some level of common
>> understanding, I won't. Instead, I'll just ask: is that the primary goal
>> here?
> 
> For me it is.  
> I do also have a need to actually learn what the next-hop's 
> supported extensions are (and I do mean "box-level" ones), 
> but that's not the main driver use-case for me at this instant in time.  

Well, I would put that into somewhat the same bucket:
You are looking to predict which options will be supported for requests 
you might send in the future.

You seem to be making an assumption that extension support is uniquely 
defined at the "box-level". I don't think its proper to assume that in 
general, for a variety of reasons. For instance, a B2BUA may choose not 
to admit to support for an extension in the context of a dialog unless 
the "other leg" supports that extension.

This goes double for other "attributes" that may be reported by OPTIONS, 
such as: event packages, mime types, media formats, even methods.

(For the latter, what is a B2BUA supposed to do if it understands some 
method, but the other "side" doesn't?)

> So for me, the main use-case for sending OPTIONS right now is to 
> learn the capability of the next-hop to process messages.  

You mean the dynamic, current "capacity" to process messages, not the 
theoretical "ability" to do so when resources are available? Or do you 
mean both?

Do you mean:
- the theoretical "ability" to do so when resources are available,
- the "capacity" to do so given current load and resource availability,
- the "willingness" to do so, at this time, for this requestor?

> Load-related state, admin state, and actual transport layer 
> reachability as well.  
> That's what many/most people have been doing with OPTIONS for 
> the past many years, after all. 

I expect that people have had one, or more, of the meanings I list above 
in mind, perhaps without realizing that they are different. I'm willing 
to bet not everyone will agree its the same one of the above. OPTIONS, 
as currently defined, cannot provide all three answers. Nor will 
tightening up the specs change that, though doing so might make it do a 
better job of *one* of those.

So one thing we might do is narrow down which (possibly more than one) 
of those we want answers to. (I expect there will be desire for all three.)

Then, lets talk about what range of things a given response applies to:

Suppose I send OPTIONS to
- sip:198.51.100.101:5060
Does the response I get apply also to
- sip:198.51.100.101:5061,
- sip:alice@198.51.100.101:5060,
- sip:alice@198.51.100.101:5060,
- sips:carol@198.51.100.101:5061
- sip:example.com	(maps to 198.51.100.101)
- sip:pbx1.example.com	(also maps to 198.51.100.101)

It *might*, or it might not.

Paul J assumes that sip:<ip-address> is "special", in that it applies to 
the "box", whereas sip:<name>@<ip-address> might return something else.
And that might make sense for a proxy, or a proxy/registrar.

But for a "phone" sip:198.51.100.101:5060 might just denote the "primary 
line", while sip:2@198.51.100.101:5060 may denote the "secondary line". 
And then the answer for one might have little predictive value over 
requests to the other. E.g. an OPTIONS to one might return 486 because 
the phone is busy, while an OPTIONS to the other might return 200.

For a gateway, there may well be DSP resources required to establish a 
call, and how much, or what type may depend on the codecs used in the 
call. So an answer to OPTIONS won't be valuable in predicting if a 
particular call can succeed.

So, I think we need to get specific about what we want to predict before 
deciding how to request the info to base those predictions on.

	Thanks,
	Paul

From HKaplan@acmepacket.com  Tue May 11 11:41:34 2010
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To: Paul Kyzivat <pkyzivat@cisco.com>
Date: Tue, 11 May 2010 14:40:57 -0400
Thread-Topic: [sipcore] The predictive value of OPTIONS
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Cc: "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] The predictive value of OPTIONS
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> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behal=
f
> Of Paul Kyzivat
> Sent: Tuesday, May 11, 2010 1:51 PM
>=20
> >
> > For me it is.
> > I do also have a need to actually learn what the next-hop's
> > supported extensions are (and I do mean "box-level" ones),
> > but that's not the main driver use-case for me at this instant in time.
>=20
> Well, I would put that into somewhat the same bucket:
> You are looking to predict which options will be supported for requests
> you might send in the future.
>=20
> You seem to be making an assumption that extension support is uniquely
> defined at the "box-level". I don't think its proper to assume that in
> general, for a variety of reasons. For instance, a B2BUA may choose not
> to admit to support for an extension in the context of a dialog unless
> the "other leg" supports that extension.
> This goes double for other "attributes" that may be reported by OPTIONS,
> such as: event packages, mime types, media formats, even methods.

Yup, I know that quite well, and have made that argument before in IETF. :)

When I said "box-level", I meant capabilities *of* the B2BUA, at least with=
 respect to me.  Not of any other "side", but of "my side".  In practice, t=
here is a set of capabilities that could be indicated, and are.  Some of th=
em would be quite useful, and really only apply on a per-hop basis anyway. =
 It's not everything, which makes it dangerous (and why I'm not suggesting =
doing it right now), but that's for a later time.

=20
> (For the latter, what is a B2BUA supposed to do if it understands some
> method, but the other "side" doesn't?)

If it accepts the method on my "side", it accepts the method.  If the other=
 side doesn't, then it will reject it when I send it.  Those two cases are =
not mutually inconsistent.  The B2BUA can make a routing decision which of =
many "sides" to send the request to, but it really does accept it and I rea=
lly should send it to find out - as opposed to when it does not accept it o=
n my side and there's no point ever sending it such a method. =20

=20
> > So for me, the main use-case for sending OPTIONS right now is to
> > learn the capability of the next-hop to process messages.
>=20
> You mean the dynamic, current "capacity" to process messages, not the
> theoretical "ability" to do so when resources are available? Or do you
> mean both?

Current.


> Do you mean:
> - the theoretical "ability" to do so when resources are available,
> - the "capacity" to do so given current load and resource availability,
> - the "willingness" to do so, at this time, for this requestor?

If you mean "for this requestor" to mean for the same hop (as opposed to fo=
r the same source Contact URI), then I'd argue the second and third are eff=
ectively the same thing, in practice.  I believe that has even come up in t=
he overload discussions a long time ago, and I think there was agreement th=
ey are effectively the same thing, logically.  Just because a next-hop tell=
s me it can't do any more, but tells someone else something different, does=
n't matter to my local processing of future requests to it.


> I expect that people have had one, or more, of the meanings I list above
> in mind, perhaps without realizing that they are different. I'm willing
> to bet not everyone will agree its the same one of the above. OPTIONS,
> as currently defined, cannot provide all three answers. Nor will
> tightening up the specs change that, though doing so might make it do a
> better job of *one* of those.

In every case I have seen this used for in deployment, as far as I can tell=
 they have actually meant the same thing.

=20
> Then, lets talk about what range of things a given response applies to:
>=20
> Suppose I send OPTIONS to
> - sip:198.51.100.101:5060
> Does the response I get apply also to
> - sip:198.51.100.101:5061,
> - sip:alice@198.51.100.101:5060,
> - sip:alice@198.51.100.101:5060,
> - sips:carol@198.51.100.101:5061
> - sip:example.com	(maps to 198.51.100.101)
> - sip:pbx1.example.com	(also maps to 198.51.100.101)

YES.  If the responder understands the Header, then it's reporting a system=
-level capability. (luckily you forgot about anycasting ;)


> But for a "phone" sip:198.51.100.101:5060 might just denote the "primary
> line", while sip:2@198.51.100.101:5060 may denote the "secondary line".
> And then the answer for one might have little predictive value over
> requests to the other. E.g. an OPTIONS to one might return 486 because
> the phone is busy, while an OPTIONS to the other might return 200.

Hence the need/desire for a Header.

=20
> For a gateway, there may well be DSP resources required to establish a
> call, and how much, or what type may depend on the codecs used in the
> call. So an answer to OPTIONS won't be valuable in predicting if a
> particular call can succeed.

That's OK.  A "box-wide" semantic is a superset.  If you can handle ANY mor=
e calls (from that previous hop), then "box-wide" you can handle more calls=
.  That doesn't mean you can't reject future calls due to unavailable resou=
rces such as DSPs.  If you can NOT handle any more "box-wide", for example =
due to CPU, then the number of DSP resources you have really doesn't matter=
.

-hadriel

From adam@nostrum.com  Tue May 11 12:13:09 2010
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Subject: [sipcore] Draft Publication Requests
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[as chair]

The SIPCORE chairs have requested publication for  the following three 
documents:

   draft-ietf-sipcore-reinvite-04
   draft-ietf-sipcore-event-rate-control-03
   draft-ietf-sipcore-invfix-01

You can track their progress using the datatracker tool:

   https://datatracker.ietf.org/doc/draft-ietf-sipcore-reinvite/
   https://datatracker.ietf.org/doc/draft-ietf-sipcore-event-rate-control/
   https://datatracker.ietf.org/doc/draft-ietf-sipcore-invfix/

/a

From HKaplan@acmepacket.com  Tue May 11 13:36:20 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Brett Tate <brett@broadsoft.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Tue, 11 May 2010 16:35:45 -0400
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> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behal=
f
> Of Brett Tate
> Sent: Tuesday, May 11, 2010 10:29 AM
>=20
> Within RFC3428 section 10 example, the proxy "finds a binding from
> sip:user2@domain.com to sip:user2@user2pc.domain.com"; however it
> indicates sip:user2@domain.com instead of sip:user2@user2pc.domain.com
> within proxied MESSAGE's request-uri.
>=20
> I assume that this a bug within the example (similar to tag and extraneou=
s
> ';' issues); however I thought that I'd ask before opening errata... is
> such proxy behavior intentional?  If intentional, what is the
> corresponding normative text within RFC3428?

It's pretty clearly a bug. :)

The text in section 6 says:
   Proxies route MESSAGE requests according to the rules of SIP [1].

And the example says:
   User1 forwards this message to the server for domain.com.  The proxy
   receives this request, and recognizes that it is the server for
   domain.com.  It looks up user2 in its database (built up through
   registrations), and finds a binding from sip:user2@domain.com to
   sip:user2@user2pc.domain.com.  It forwards the request to user2.

So F2 should have been:
   MESSAGE sip:user2@user2pc.domain.com SIP/2.0
   Via: SIP/2.0/TCP proxy.domain.com;branch=3Dz9hG4bK123dsghds
   Via: SIP/2.0/TCP user1pc.domain.com;branch=3Dz9hG4bK776sgdkse;
                                              received=3D1.2.3.4
   Max-Forwards: 69
   From: sip:user1@domain.com;tag=3D49394
   To: sip:user2@domain.com
   Call-ID: asd88asd77a@1.2.3.4
   CSeq: 1 MESSAGE
   Content-Type: text/plain
   Content-Length: ...
   ...

-hadriel

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On May 10, 2010, at 10:55 PM, Hadriel Kaplan wrote:

>  Paul is trying to alleviate those interop issues by recommending a =
particular configuration, which follows rfc3261's language.  One could =
almost call it a BCP. (except the term "BCP" means something different =
in IETF than normal nomenclature)
>=20
> So he submitted a draft to do that.  The response he gets is "we're =
working on an overload control mechanism and this thing has a bit of =
that... so no, we can't clean this up".


Really? I did not get that out of the draft. If you mean that the drafts =
thinks it is OK to a SIP URI that has a host portion with no user name, =
well I took that as fairly obvious that was OK in 3261.  Section 11 =
explicitly discusses it.=20






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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Cullen Jennings <fluffy@cisco.com>
Date: Tue, 11 May 2010 17:30:13 -0400
Thread-Topic: [sipcore] draft-jones-sip-options-ping-01
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Cc: "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] draft-jones-sip-options-ping-01
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> -----Original Message-----
> From: Cullen Jennings [mailto:fluffy@cisco.com]
> Sent: Tuesday, May 11, 2010 5:15 PM
> To: Hadriel Kaplan
>=20
> On May 10, 2010, at 10:55 PM, Hadriel Kaplan wrote:
>=20
> >  Paul is trying to alleviate those interop issues by recommending a
> particular configuration, which follows rfc3261's language.  One could
> almost call it a BCP. (except the term "BCP" means something different in
> IETF than normal nomenclature)
> > So he submitted a draft to do that.  The response he gets is "we're
> working on an overload control mechanism and this thing has a bit of
> that... so no, we can't clean this up".
>=20
> Really? I did not get that out of the draft. If you mean that the drafts
> thinks it is OK to a SIP URI that has a host portion with no user name,
> well I took that as fairly obvious that was OK in 3261.  Section 11
> explicitly discusses it.

Right, but it wasn't clear enough to make it usable without configuration i=
n many cases.  First, because if you send it to a proxy or b2bua between yo=
u and the Registrar, it's not distinguishable from an OPTIONS targeted to t=
he Registrar, so some proxies/b2buas will forward it to the registrar.  If =
you set a max-forwards of 0 to stop that, you'll get a 483 from some proxie=
s/b2buas instead of a OPTIONS-based response.  That's legit too, since 3261=
 only said:
   Alternatively, a server receiving an OPTIONS request with a Max-
   Forwards header field value of 0 MAY respond to the request
   regardless of the Request-URI.

Then there's the response status code itself and clarifying section 11.2.

So afaict Paul's draft is trying to clear up 3261's language to be explicit=
 and avoid the manual configuration for these types of issues.  (though I m=
ay be reading more into it than is there, or than Paul intends)

-hadriel

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Cc: "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] The predictive value of OPTIONS
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This is fragmenting into multiple issues that ought to be discussed 
separately. But I want to go home so I'm not splitting it up.

inline...

Hadriel Kaplan wrote:
> 
>> -----Original Message-----
>> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf
>> Of Paul Kyzivat
>> Sent: Tuesday, May 11, 2010 1:51 PM
>>
>>> For me it is.
>>> I do also have a need to actually learn what the next-hop's
>>> supported extensions are (and I do mean "box-level" ones),
>>> but that's not the main driver use-case for me at this instant in time.
>> Well, I would put that into somewhat the same bucket:
>> You are looking to predict which options will be supported for requests
>> you might send in the future.
>>
>> You seem to be making an assumption that extension support is uniquely
>> defined at the "box-level". I don't think its proper to assume that in
>> general, for a variety of reasons. For instance, a B2BUA may choose not
>> to admit to support for an extension in the context of a dialog unless
>> the "other leg" supports that extension.
>> This goes double for other "attributes" that may be reported by OPTIONS,
>> such as: event packages, mime types, media formats, even methods.
> 
> Yup, I know that quite well, and have made that argument before in IETF. :)
> 
> When I said "box-level", I meant capabilities *of* the B2BUA, at least with respect to me.  Not of any other "side", but of "my side".  In practice, there is a set of capabilities that could be indicated, and are.  Some of them would be quite useful, and really only apply on a per-hop basis anyway.  It's not everything, which makes it dangerous (and why I'm not suggesting doing it right now), but that's for a later time.
> 
>  
>> (For the latter, what is a B2BUA supposed to do if it understands some
>> method, but the other "side" doesn't?)
> 
> If it accepts the method on my "side", it accepts the method.  If the other side doesn't, then it will reject it when I send it.  Those two cases are not mutually inconsistent.  The B2BUA can make a routing decision which of many "sides" to send the request to, but it really does accept it and I really should send it to find out - as opposed to when it does not accept it on my side and there's no point ever sending it such a method.  

Lets use MESSAGE for discussion. You would report MESSAGE in an Accept 
header. Then later you receive MESSAGE, forward it to the "other side", 
and get back a 405, including a list of methods that are supported, not 
including MESSAGE. Do you pass that Accept on? Or alter it to indicate 
that you *do* support MESSAGE even though you are rejecting it with a 405?

There is no way to win here. You must be inconsistent at some point. So 
the result of the OPTIONS wrt Accept cannot be trusted to predict future 
behavior. The same is true for most of the other things reported in 
OPTIONS - its really hard to say much about how relevant they are for 
predicting future behavior.

To do better, I think it is necessary to either:
- specify in the query the context of the question, or
- return a complex document that describes what is supported under
   a variety of circumstances.

I doubt we want the latter (at least I don't).

Specifying the context for which you want an answer is also pretty 
difficult to do, given the number of different contexts that are 
possible. Maybe its possible to do a few interesting ones.

One is to send an OPTIONS to a URL and get an answer that pertains to 
*that* URL, while not necessarily pertaining to any other URL. AFAIK 
that is, more or less, what the current definition of OPTIONS does. You 
might choose to extrapolate the results for that URL to other URLs, but 
that may or may not produce correct results. Of course this is also 
limited, in that it only reports the results for OPTIONS. It has no way 
to customize the results for other methods. (For instance, some MIME 
types are likely to be acceptable only in the context of certain 
methods. Application/SDP may not be acceptable for MESSAGE.)

To get beyond the answer for one URL, we need a way to indicate we want 
an answer for a class of URLs, maybe all of them. Some seem to think 
that a URL with no user part is an indicator for this. I think that is 
probably a sensible meaning for some servers, but not all of them. IMO 
there is no good value you can pick to put in the user part of the R-URI 
to indicate this. But perhaps it could be indicated via some new header. 
If this is the desire, and single answers are desired, then some 
accuracy must be given up. Presumably the answer must be the union of 
the answers for all the URIs it applies to. So for any particular URI 
the answer might be less.

Also, there is the distinction between what might be available on a good 
day, vs. what the server is currently capable of, vs. what it is 
*willing* to do. Again, I don't see how any one answer is the right one 
for this. (E.g. the server is heavily loaded. It is capable of accepting 
*some* requests, but is only willing to accept requests from privileged 
callers. Or, it is in shutting-down mode. It has plenty of resources, 
but is only willing to accept in-dialog requests.)

Again, I think to get at this will require an indicator in the request 
of which sort of answer you want. Unless we can agree the answer is 
always and only about *capability*, and never about policy.

Bottom line here: I think there needs to be a selector, or selectors, 
for which kind of answer is desired.

>>> So for me, the main use-case for sending OPTIONS right now is to
>>> learn the capability of the next-hop to process messages.
>> You mean the dynamic, current "capacity" to process messages, not the
>> theoretical "ability" to do so when resources are available? Or do you
>> mean both?
> 
> Current.

Sorry - I botched my message that you are replying to. I wrote the 
above, then the list immediately below to replace that.

>> Do you mean:
>> - the theoretical "ability" to do so when resources are available,
>> - the "capacity" to do so given current load and resource availability,
>> - the "willingness" to do so, at this time, for this requestor?
> 
> If you mean "for this requestor" to mean for the same hop (as opposed to for the same source Contact URI), then I'd argue the second and third are effectively the same thing, in practice.  I believe that has even come up in the overload discussions a long time ago, and I think there was agreement they are effectively the same thing, logically.  Just because a next-hop tells me it can't do any more, but tells someone else something different, doesn't matter to my local processing of future requests to it.

I meant whatever the server would use as input to any policy decisions 
it might make. That might mean the sender of the message (top Via 
value), the originator of the request, whatever.

For instance, administrative shutdown would be a policy that determines 
willingness to handle requests. (At least until the listeners are 
stopped.) It might apply to certain links, which is what you are getting 
at. It might also apply to certain request originators, or to in-dialog 
vs. out-of-dialog requests, etc. It might also apply to certain R-URIs.

Policies to throttle load before overload occurs could also fall into 
this category of "willingness".

The specified expectation that an OPTIONS should return 486 if the 
device (a phone presumably) is on a call also falls into this "policy" 
category, since it could alternatively hang up the existing call or put 
it on hold to accept the new call.

I can see utility in getting theoretical capability as well: it would be 
nice to be able to determine which if any of your devices could support 
video, even if currently busy. I might then choose to wait till the 
video phone is free rather than calling now on the audio phone. (But of 
course this is better done with presence.)

But, if the majority of people can agree on a single context in which 
they want an answer, then perhaps we can simply have a single indicator 
that says we want that, rather than whatever it is that OPTIONS 
otherwise returns.

>> I expect that people have had one, or more, of the meanings I list above
>> in mind, perhaps without realizing that they are different. I'm willing
>> to bet not everyone will agree its the same one of the above. OPTIONS,
>> as currently defined, cannot provide all three answers. Nor will
>> tightening up the specs change that, though doing so might make it do a
>> better job of *one* of those.
> 
> In every case I have seen this used for in deployment, as far as I can tell they have actually meant the same thing.

To be fair, ACME has a particular perspective. SBCs talk to a certain 
class of equipment. I doubt you have need to interface directly to 
phones. But others do use OPTIONS to phones.

>> Then, lets talk about what range of things a given response applies to:
>>
>> Suppose I send OPTIONS to
>> - sip:198.51.100.101:5060
>> Does the response I get apply also to
>> - sip:198.51.100.101:5061,
>> - sip:alice@198.51.100.101:5060,
>> - sip:alice@198.51.100.101:5060,
>> - sips:carol@198.51.100.101:5061
>> - sip:example.com	(maps to 198.51.100.101)
>> - sip:pbx1.example.com	(also maps to 198.51.100.101)
> 
> YES.  If the responder understands the Header, then it's reporting a system-level capability. (luckily you forgot about anycasting ;)

I slipped some ringers in there for you. :-)

There is no guarantee that two ports for the same ip address are served 
by the same "server". They may be served by separate processes with 
independent resources. Statistically there is probably a good chance 
that 5060 and 5061 are served by the same server. But once you get past 
that, then what?

I didn't put in multi-homing situations, or IPv6/IPv6 cases, but they 
apply too. For all of these I can't see a good alternative to assuming 
they are distinct and asking on each, or *configuring* the cases when 
you *know* they are the same.

And I still disagree with assuming they are the same when the user part 
varies, unless we add something to the message to indicate that is what 
is desired. But even an indicator won't deal with the port issue, or 
with multihoming, since you can't know if the device speaks for the 
others. OR, the response could be altered to indicate all the different 
addresses is applies to.

>> But for a "phone" sip:198.51.100.101:5060 might just denote the "primary
>> line", while sip:2@198.51.100.101:5060 may denote the "secondary line".
>> And then the answer for one might have little predictive value over
>> requests to the other. E.g. an OPTIONS to one might return 486 because
>> the phone is busy, while an OPTIONS to the other might return 200.
> 
> Hence the need/desire for a Header.
> 
>  
>> For a gateway, there may well be DSP resources required to establish a
>> call, and how much, or what type may depend on the codecs used in the
>> call. So an answer to OPTIONS won't be valuable in predicting if a
>> particular call can succeed.
> 
> That's OK.  A "box-wide" semantic is a superset.  If you can handle ANY more calls (from that previous hop), then "box-wide" you can handle more calls.  That doesn't mean you can't reject future calls due to unavailable resources such as DSPs.  If you can NOT handle any more "box-wide", for example due to CPU, then the number of DSP resources you have really doesn't matter.

That is a possible interpretation.

If some (expensive) codecs are disabled during peak load, should that be 
reflected in the response?

What if some destinations are currently unreachable? Should OPTIONS for 
a particular address reflect that? (E.g. only +1... calls are currently 
operational because the international trunk is down. Should OPTIONS to 
sip:+44... return 480 or 200?)

	Thanks,
	Paul

From HKaplan@acmepacket.com  Tue May 11 19:39:20 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
Date: Tue, 11 May 2010 22:39:00 -0400
Thread-Topic: [sipcore] The predictive value of OPTIONS
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Subject: Re: [sipcore] The predictive value of OPTIONS
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> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Tuesday, May 11, 2010 8:40 PM
> To: Hadriel Kaplan
>=20
> Hadriel Kaplan wrote:
> >
> >> (For the latter, what is a B2BUA supposed to do if it understands some
> >> method, but the other "side" doesn't?)
> >
> > If it accepts the method on my "side", it accepts the method.  If the
> other side doesn't, then it will reject it when I send it.  Those two
> cases are not mutually inconsistent.  The B2BUA can make a routing
> decision which of many "sides" to send the request to, but it really does
> accept it and I really should send it to find out - as opposed to when it
> does not accept it on my side and there's no point ever sending it such a
> method.
>=20
> Lets use MESSAGE for discussion. You would report MESSAGE in an Accept
> header. Then later you receive MESSAGE, forward it to the "other side",
> and get back a 405, including a list of methods that are supported, not
> including MESSAGE. Do you pass that Accept on? Or alter it to indicate
> that you *do* support MESSAGE even though you are rejecting it with a 405=
?
>=20
> There is no way to win here. You must be inconsistent at some point. So
> the result of the OPTIONS wrt Accept cannot be trusted to predict future
> behavior. The same is true for most of the other things reported in
> OPTIONS - its really hard to say much about how relevant they are for
> predicting future behavior.

I think you're missing my point - when you send a request to user1@foo what=
 you get back is about user1@foo.  So if you get back an Accept without MES=
SAGE, you may be able to know something about user1@foo, but NOTHING about =
user2@foo.  There is no inconsistency there.

But as I said in the previous email with regards to what could reasonably b=
e learned from the response: "It's not everything, which makes it dangerous=
 (and why I'm not suggesting doing it right now), but that's for a later ti=
me."  Allowed methods, for example, is probably a borderline one.  But most=
 of the capabilities which I would want such a thing for would really be pr=
oxy type of capabilities: things like keepalives, sigcomp, secagree, and so=
me transport/connection things which should arguably be in DNS SRV's but ar=
e basically always manually provisioned today.

=20
> One is to send an OPTIONS to a URL and get an answer that pertains to
> *that* URL, while not necessarily pertaining to any other URL. AFAIK
> that is, more or less, what the current definition of OPTIONS does. You
> might choose to extrapolate the results for that URL to other URLs, but
> that may or may not produce correct results. Of course this is also
> limited, in that it only reports the results for OPTIONS. It has no way
> to customize the results for other methods. (For instance, some MIME
> types are likely to be acceptable only in the context of certain
> methods. Application/SDP may not be acceptable for MESSAGE.)
>=20
> To get beyond the answer for one URL, we need a way to indicate we want
> an answer for a class of URLs, maybe all of them. Some seem to think
> that a URL with no user part is an indicator for this. I think that is
> probably a sensible meaning for some servers, but not all of them. IMO
> there is no good value you can pick to put in the user part of the R-URI
> to indicate this. But perhaps it could be indicated via some new header.
> If this is the desire, and single answers are desired, then some
> accuracy must be given up. Presumably the answer must be the union of
> the answers for all the URIs it applies to. So for any particular URI
> the answer might be less.

Exactly - when you send it to a URL, the result is about the URL; and curre=
ntly lots of devices do assume no user part is effectively all of them (of =
that particular URL), and I have been agreeing that a Header is a fine way =
to make it unambiguous for that case.  I don't think anyone has objected to=
 that, including Paul J.

=20
> Also, there is the distinction between what might be available on a good
> day, vs. what the server is currently capable of, vs. what it is
> *willing* to do. Again, I don't see how any one answer is the right one
> for this. (E.g. the server is heavily loaded. It is capable of accepting
> *some* requests, but is only willing to accept requests from privileged
> callers. Or, it is in shutting-down mode. It has plenty of resources,
> but is only willing to accept in-dialog requests.)
>=20
> Again, I think to get at this will require an indicator in the request
> of which sort of answer you want. Unless we can agree the answer is
> always and only about *capability*, and never about policy.
>=20
> Bottom line here: I think there needs to be a selector, or selectors,
> for which kind of answer is desired.

I don't think so.  Not in practice.  There is no real difference to the req=
uestor whether the responder won't accept new requests because it's overloa=
ded, shutting down, or just doesn't like you.  The resulting effect on the =
requestor is the same.  And at the end of the day, that's the important pie=
ce for interoperability: what the requestor should do given a response, not=
 why the responder sent the response.

For the "privileged callers" case, if by "privileged callers" you mean spec=
ific calls from the same hop, then the responder really does accept request=
s, just not all requests, and should say it does accept requests - but that=
's *OK*, it's not _broken_ to not be able to signal every possible policy r=
ule to previous hops.  It may not be optimal in some folks' minds, but it's=
 not broken.  Same goes for the other "policy" cases you had in your email =
(I've removed them to reduce the size of this email).

=20
> > In every case I have seen this used for in deployment, as far as I can
> tell they have actually meant the same thing.
>=20
> To be fair, ACME has a particular perspective. SBCs talk to a certain
> class of equipment. I doubt you have need to interface directly to
> phones. But others do use OPTIONS to phones.

Actually over half of our SBC's talk to phones direct. (I count an MTA/IAD/=
Terminal as a phone - basically anything that Registers and represents a si=
ngle user I consider a "phone")  But you're right SBCs don't generally send=
 OPTIONS to phones unless something else sends it through the SBC to reach =
them, and I don't see that happening very much.  I usually see OPTIONS comi=
ng from phones to the SBC, or between an SBC and its next/previous-hop serv=
ers.

=20
> >> Then, lets talk about what range of things a given response applies to=
:
> >>
> >> Suppose I send OPTIONS to
> >> - sip:198.51.100.101:5060
> >> Does the response I get apply also to
> >> - sip:198.51.100.101:5061,
> >> - sip:alice@198.51.100.101:5060,
> >> - sip:alice@198.51.100.101:5060,
> >> - sips:carol@198.51.100.101:5061
> >> - sip:example.com	(maps to 198.51.100.101)
> >> - sip:pbx1.example.com	(also maps to 198.51.100.101)
> >
> > YES.  If the responder understands the Header, then it's reporting a
> system-level capability. (luckily you forgot about anycasting ;)
>=20
> I slipped some ringers in there for you. :-)

Oops, missed the port numbers being different.  Right, those are different,=
 because they're a different URL hostport.

-hadriel

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To: "Hadriel Kaplan" <HKaplan@acmepacket.com>
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Subject: Re: [sipcore] draft-jones-sip-options-ping-01
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Hmm, I'm still confused what the problem is that causes me to want to =
send an options to say my outbound proxy but if my outbound =
proxy/sbc/b2bua is configured to be outbound.example.com and my AOR it =
alice@example.com, it seems pretty clear that I send it to =
sip:outbound.example.com if that's where I want it to go.=20


On May 11, 2010, at 3:30 PM, Hadriel Kaplan wrote:

>=20
>=20
> > -----Original Message-----
> > From: Cullen Jennings [mailto:fluffy@cisco.com]
> > Sent: Tuesday, May 11, 2010 5:15 PM
> > To: Hadriel Kaplan
> >
> > On May 10, 2010, at 10:55 PM, Hadriel Kaplan wrote:
> >
> > >  Paul is trying to alleviate those interop issues by recommending =
a
> > particular configuration, which follows rfc3261's language.  One =
could
> > almost call it a BCP. (except the term "BCP" means something =
different in
> > IETF than normal nomenclature)
> > > So he submitted a draft to do that.  The response he gets is =
"we're
> > working on an overload control mechanism and this thing has a bit of
> > that... so no, we can't clean this up".
> >
> > Really? I did not get that out of the draft. If you mean that the =
drafts
> > thinks it is OK to a SIP URI that has a host portion with no user =
name,
> > well I took that as fairly obvious that was OK in 3261.  Section 11
> > explicitly discusses it.
>=20
> Right, but it wasn't clear enough to make it usable without =
configuration in many cases.  First, because if you send it to a proxy =
or b2bua between you and the Registrar, it's not distinguishable from an =
OPTIONS targeted to the Registrar, so some proxies/b2buas will forward =
it to the registrar.  If you set a max-forwards of 0 to stop that, =
you'll get a 483 from some proxies/b2buas instead of a OPTIONS-based =
response.  That's legit too, since 3261 only said:
>    Alternatively, a server receiving an OPTIONS request with a Max-
>    Forwards header field value of 0 MAY respond to the request
>    regardless of the Request-URI.
>=20
> Then there's the response status code itself and clarifying section =
11.2.
>=20
> So afaict Paul's draft is trying to clear up 3261's language to be =
explicit and avoid the manual configuration for these types of issues.  =
(though I may be reading more into it than is there, or than Paul =
intends)
>=20
> -hadriel
>=20


Cullen Jennings
For corporate legal information go to:
http://www.cisco.com/web/about/doing_business/legal/cri/index.html




From shinji.okumura@softfront.jp  Wed May 12 04:13:25 2010
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From: OKUMURA Shinji <shinji.okumura@softfront.jp>
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Hi,

I have one question and one comment about UAC behavior.

question:
At first, oa status is only audio, and during a next reINVITE
transaction oa status is changed to audio and video. And then
UAS sends a 4xx response to the reINVITE.
In this case which is a new offer UAC should send, only audio
or both of audio and video with zero port?

comment:
If already-executed changes is made by only PRACK transaction,
IMO UAC need not generate a new offer to synchronize both UAs.

Regards,
Shinji

Internet-Drafts@ietf.org
Sat,  8 May 2010 00:30:02 -0700 (PDT)
>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-04.txt
>	Pages           : 24
>	Date            : 2010-05-08
>
>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.
>
>A URL for this Internet-Draft is:
>http://www.ietf.org/internet-drafts/draft-ietf-sipcore-reinvite-04.txt
>
>Internet-Drafts are also available by anonymous FTP at:
>ftp://ftp.ietf.org/internet-drafts/
>
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>Internet-Draft.

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> Hi,
> 
> I have one question and one comment about UAC behavior.
> 
> question:
> At first, oa status is only audio, and during a next reINVITE
> transaction oa status is changed to audio and video. And then
> UAS sends a 4xx response to the reINVITE.

If the O/A in Re-INVITE without precondition, the media modification is 
defined as *in use* after the O/A by RFC3264.
So, the UAS need to send 2xx, while not 4xx.
 
> In this case which is a new offer UAC should send, only audio
> or both of audio and video with zero port?

Here, if you are talking the *legacy* UAS, then the *legacy* UAS may send 
4xx.
This case is talked in "3.4.  UAC Behavior" of the text.

Current normative text is that UAC SHOULD send a new Offer to synchronize 
the session state. As the session state would be synchronize by the new 
O/A, so there's no rigorous rule about the content of the Offer. But the 
text show tendency by the description of:
the UAC uses the criteria in Section 3.3 in order to decide whether or not 
changes have been executed for the stream

I guess this is the deliberate arrangement by Gonzalo. Gonzalo?

> 
> comment:
> If already-executed changes is made by only PRACK transaction,
> IMO UAC need not generate a new offer to synchronize both UAs.

New Offer is used by UAC in the conditions:
"modification is *in use*, AND UAS send 4xx" or other race condition.

I think it is not relating to the request type.

Thanks,

Gao


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<br><tt><font size=2><br>
&gt; Hi,<br>
&gt; <br>
&gt; I have one question and one comment about UAC behavior.<br>
&gt; <br>
&gt; question:<br>
&gt; At first, oa status is only audio, and during a next reINVITE<br>
&gt; transaction oa status is changed to audio and video. And then<br>
&gt; UAS sends a 4xx response to the reINVITE.</font></tt>
<br>
<br><tt><font size=2>If the O/A in Re-INVITE without precondition, the
media modification is defined as *in use* after the O/A by RFC3264.</font></tt>
<br><tt><font size=2>So, the UAS need to send 2xx, while not 4xx.</font></tt>
<br><tt><font size=2>&nbsp;<br>
&gt; In this case which is a new offer UAC should send, only audio<br>
&gt; or both of audio and video with zero port?</font></tt>
<br>
<br><tt><font size=2>Here, if you are talking the *legacy* UAS, then the
*legacy* UAS may send 4xx.</font></tt>
<br><tt><font size=2>This case is talked in &quot;3.4. &nbsp;UAC Behavior&quot;
of the text.</font></tt>
<br>
<br><tt><font size=2>Current normative text is that UAC SHOULD send a new
Offer to synchronize the session state. As the session state would be synchronize
by the new O/A, so there's no rigorous rule about the content of the Offer.
But the text show tendency by the description of:</font></tt>
<br><tt><font size=2>the UAC uses the criteria in Section 3.3 in order
to decide whether or not changes have been executed for the stream</font></tt>
<br>
<br><tt><font size=2>I guess this is the deliberate arrangement by Gonzalo.
Gonzalo?</font></tt>
<br><tt><font size=2><br>
&gt; <br>
&gt; comment:<br>
&gt; If already-executed changes is made by only PRACK transaction,<br>
&gt; IMO UAC need not generate a new offer to synchronize both UAs.</font></tt>
<br>
<br><tt><font size=2>New Offer is used by UAC in the conditions:</font></tt>
<br><tt><font size=2>&quot;modification is *in use*, AND UAS send 4xx&quot;
or other race condition.</font></tt>
<br>
<br><tt><font size=2>I think it is not relating to the request type.</font></tt>
<br>
<br><tt><font size=2>Thanks,</font></tt>
<br>
<br><tt><font size=2>Gao</font></tt>
<br><br><pre>
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From dworley@avaya.com  Wed May 12 13:13:58 2010
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From: "WORLEY, Dale R (Dale)" <dworley@avaya.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>, Paul Kyzivat <pkyzivat@cisco.com>
Date: Wed, 12 May 2010 16:13:40 -0400
Thread-Topic: [sipcore] The predictive value of OPTIONS
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References: <01c501caec0e$48e624e0$dab26ea0$@com> <FF84A09F50A6DC48ACB6714F4666CC74632B2DA14B@ESESSCMS0354.eemea.ericsson.se> <430FC6BDED356B4C8498F634416644A91A8A7D7192@mail> <FF84A09F50A6DC48ACB6714F4666CC74632B2DA963@ESESSCMS0354.eemea.ericsson.se> <430FC6BDED356B4C8498F634416644A91A8A7D71A2@mail>	<4BE2BCB0.10909@cisco.com> <430FC6BDED356B4C8498F634416644A91A8A7D760B@mail> <4BE44683.9040405@cisco.com> <430FC6BDED356B4C8498F634416644A91A8A7D7771@mail> <4BE81570.5080803@cisco.com>	<f711ff46b5a6.4be7e1eb@us.army.mil> <00d901caf076$bdd21770$39764650$@com> <4BE86339.6020709@cisco.com> <010501caf087$53ff6c60$fbfe4520$@com> <4BE889BD.2030901@cisco.com> <430FC6BDED356B4C8498F634416644A91A8A7D7AB9@mail> <4BE95E6E.3000601@cisco.com> <430FC6BDED356B4C8498F634416644A91C139909C2@mail> <4BE998FE.5040703@cisco.com> <430FC6BDED356B4C8498F634416644A91C13990ABF@mail> <4BE9F8CE.2050207@cisco.com>, <430FC6BDED356B4C8498F634416644A91C13990BE6@mail>
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Subject: Re: [sipcore] The predictive value of OPTIONS
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There is some predictive value of a blanket OPTIONS test, probably more tha=
n people realize.

In SIP, there can be multiple ways to accomplish a task.  Especially when p=
ropprietary extensions may be present, the theoretically correct way to han=
dle the situation is to try the methods in order of decreasing preference u=
ntil you discover a method that is supported:  Try A, if that fails, try B,=
 if that fails, try C.  In practice, you'd like to know if the next-hop ele=
ment doesn't support method A, so you can avoid the overhead of trying it o=
n future calls.

In the case of proprietary extensions (which method A is likely to be), a s=
ingle OPTIONs request can tell you if the element has never heard of the ex=
tension.  Because of downstream elements, you may still have to fall back t=
o methods B and C at times, but there is a significant savings if you can d=
etect the cases where the next-hop element cannot process method A *at all*=
.

Dale


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OKUMURA Shinji wrote:
> Hi,
> 
> I have one question and one comment about UAC behavior.
> 
> question:
> At first, oa status is only audio, and during a next reINVITE
> transaction oa status is changed to audio and video. And then
> UAS sends a 4xx response to the reINVITE.
> In this case which is a new offer UAC should send, only audio
> or both of audio and video with zero port?

Well, the new o/a is to restate the status quo.
Since the new o/a never succeeded, the status quo includes only one 
m-line. So I think it would be acceptable to send an offer with that.

OTOH, its possible that the other party is confused and thinks there are 
two m-lines in effect. It might be further confused to receive another 
offer with only one m-line. So it might be prudent to send the new offer 
as you suggest. Its legal to do so, and it ought not hurt anything.
(But it might cause trouble if the other UA simply can't handle the 
additional m-line. If so it is quite broken.)

> comment:
> If already-executed changes is made by only PRACK transaction,
> IMO UAC need not generate a new offer to synchronize both UAs.

I don't understand what point you are making, or why.

	Thanks,
	Paul

> Regards,
> Shinji
> 
> Internet-Drafts@ietf.org
> Sat,  8 May 2010 00:30:02 -0700 (PDT)
>> 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-04.txt
>> 	Pages           : 24
>> 	Date            : 2010-05-08
>>
>> 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.
>>
>> A URL for this Internet-Draft is:
>> http://www.ietf.org/internet-drafts/draft-ietf-sipcore-reinvite-04.txt
>>
>> Internet-Drafts are also available by anonymous FTP at:
>> ftp://ftp.ietf.org/internet-drafts/
>>
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From shinji.okumura@softfront.jp  Wed May 12 18:17:26 2010
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Gao,

gao.yang2@zte.com.cn
Wed, 12 May 2010 21:10:02 +0800
>> Hi,
>> 
>> I have one question and one comment about UAC behavior.
>> 
>> question:
>> At first, oa status is only audio, and during a next reINVITE
>> transaction oa status is changed to audio and video. And then
>> UAS sends a 4xx response to the reINVITE.
>
>If the O/A in Re-INVITE without precondition, the media modification is 
>defined as *in use* after the O/A by RFC3264.
>So, the UAS need to send 2xx, while not 4xx.
> 
>> In this case which is a new offer UAC should send, only audio
>> or both of audio and video with zero port?
>
>Here, if you are talking the *legacy* UAS, then the *legacy* UAS may send 
>4xx.
>This case is talked in "3.4.  UAC Behavior" of the text.
>
>Current normative text is that UAC SHOULD send a new Offer to synchronize 
>the session state. As the session state would be synchronize by the new 
>O/A, so there's no rigorous rule about the content of the Offer. But the 
>text show tendency by the description of:
>the UAC uses the criteria in Section 3.3 in order to decide whether or not 
>changes have been executed for the stream
>
>I guess this is the deliberate arrangement by Gonzalo. Gonzalo?

My question is not so complicated.
If a UAS recognize that current oa state is audio and video.
It means SDP include two m= lines.
In this condition, may UAC send offer SDP that include only one
m= line?

Regards,
Shinji

>> comment:
>> If already-executed changes is made by only PRACK transaction,
>> IMO UAC need not generate a new offer to synchronize both UAs.
>
>New Offer is used by UAC in the conditions:
>"modification is *in use*, AND UAS send 4xx" or other race condition.
>
>I think it is not relating to the request type.
>
>Thanks,
>
>Gao

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Hi Paul and Gao,

Paul Kyzivat <pkyzivat@cisco.com>
Wed, 12 May 2010 16:49:31 -0400
>OKUMURA Shinji wrote:
>> Hi,
>> 
>> I have one question and one comment about UAC behavior.
>> 
>> question:
>> At first, oa status is only audio, and during a next reINVITE
>> transaction oa status is changed to audio and video. And then
>> UAS sends a 4xx response to the reINVITE.
>> In this case which is a new offer UAC should send, only audio
>> or both of audio and video with zero port?
>
>Well, the new o/a is to restate the status quo.
>Since the new o/a never succeeded, the status quo includes only one 
>m-line. So I think it would be acceptable to send an offer with that.
>
>OTOH, its possible that the other party is confused and thinks there are 
>two m-lines in effect. It might be further confused to receive another 
>offer with only one m-line. So it might be prudent to send the new offer 
>as you suggest. Its legal to do so, and it ought not hurt anything.
>(But it might cause trouble if the other UA simply can't handle the 
>additional m-line. If so it is quite broken.)
>
>> comment:
>> If already-executed changes is made by only PRACK transaction,
>> IMO UAC need not generate a new offer to synchronize both UAs.
>
>I don't understand what point you are making, or why.

As Reconfirmation of fundamental premise:

If reINVITE is rejected, then both UAs must cancel all of
changes within the reINVITE transaction.
Originally this rollback is an automatic process, so sending
new offer is not necessary.

Why should UAC send new offer?
The reason is that:
   "There are also certain
   race condition situations that get both user agents out of
   synchronization" as shown in Figure 5.

In Figure 5, if UPDATE transaction does not exist, the common
view of oa state is already synchronized(i.e., sendrecv audio
and inactive video).

I think in this case UAC need not generate a new offer.

Regards,
Shinji

>	Thanks,
>	Paul
>
>> Regards,
>> Shinji
>> 
>> Internet-Drafts@ietf.org
>> Sat,  8 May 2010 00:30:02 -0700 (PDT)
>>> 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-04.txt
>>> 	Pages           : 24
>>> 	Date            : 2010-05-08
>>>
>>> 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.
>>>
>>> A URL for this Internet-Draft is:
>>> http://www.ietf.org/internet-drafts/draft-ietf-sipcore-reinvite-04.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.

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Hi Shinji,

> My question is not so complicated.
> If a UAS recognize that current oa state is audio and video.
> It means SDP include two m= lines.
> In this condition, may UAC send offer SDP that include only one
> m= line?

Sure.

But I'd like to point out that this UAS is *legacy* one. As it is *legacy* 
one, the UAC need to send a new O/A to synchronize the session state. And 
the UAC can not know the UAS's *understanding* of the state(*legacy* one 
is ambiguous).

Thanks,

Gao


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<br>
<br><tt><font size=2>Hi Shinji,</font></tt>
<br><tt><font size=2><br>
&gt; My question is not so complicated.<br>
&gt; If a UAS recognize that current oa state is audio and video.<br>
&gt; It means SDP include two m= lines.<br>
&gt; In this condition, may UAC send offer SDP that include only one<br>
&gt; m= line?<br>
</font></tt>
<br><tt><font size=2>Sure.</font></tt>
<br>
<br><tt><font size=2>But I'd like to point out that this UAS is *legacy*
one. As it is *legacy* one, the UAC need to send a new O/A to synchronize
the session state. And the UAC can not know the UAS's *understanding* of
the state(*legacy* one is ambiguous).</font></tt>
<br>
<br><tt><font size=2>Thanks,</font></tt>
<br>
<br><tt><font size=2>Gao</font></tt>
<br><br><pre>
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From shinji.okumura@softfront.jp  Wed May 12 19:08:19 2010
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From: OKUMURA Shinji <shinji.okumura@softfront.jp>
To: sipcore@ietf.org
Date: Thu, 13 May 2010 11:08:00 +0900
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Subject: Re: [sipcore] I-D Action:draft-ietf-sipcore-reinvite-04.txt
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Gao,

I am talking about "UAC Behavior".
What you say is "UAS Behavior".

The basic principle in my opinion is, 

If an UPDATE has been not used, then the original rollback
process may works well.

Regards,
Shinji

gao.yang2@zte.com.cn
Thu, 13 May 2010 09:37:19 +0800
>Hi Shinji,
>
>As I said in my first mail in this thread, the UAS should not send 4xx by 
>the new rule in this draft.
>
>The basic principle in this draft is, 
>
>1. If the modification triggered by the Re-INVITE is *in use*, then the 
>UAS should not send 4xx.
>
>2. If the UAS want to rollback modification which has been *in use*, an 
>UPDATE before final 2xx of Re-INVITE is suggested.
>
>Thanks,
>
>Gao
>
>===================================
> Zip    : 210012
> Tel    : 87211
> Tel2   :(+86)-025-52877211
> e_mail : gao.yang2@zte.com.cn
>===================================
>
>OKUMURA Shinji <shinji.okumura@softfront.jp> 
>2010-05-13 09:30
>
>Hi Paul and Gao,
>
>Paul Kyzivat <pkyzivat@cisco.com>
>Wed, 12 May 2010 16:49:31 -0400
>>OKUMURA Shinji wrote:
>>> Hi,
>>> 
>>> I have one question and one comment about UAC behavior.
>>> 
>>> question:
>>> At first, oa status is only audio, and during a next reINVITE
>>> transaction oa status is changed to audio and video. And then
>>> UAS sends a 4xx response to the reINVITE.
>>> In this case which is a new offer UAC should send, only audio
>>> or both of audio and video with zero port?
>>
>>Well, the new o/a is to restate the status quo.
>>Since the new o/a never succeeded, the status quo includes only one 
>>m-line. So I think it would be acceptable to send an offer with that.
>>
>>OTOH, its possible that the other party is confused and thinks there are 
>>two m-lines in effect. It might be further confused to receive another 
>>offer with only one m-line. So it might be prudent to send the new offer 
>>as you suggest. Its legal to do so, and it ought not hurt anything.
>>(But it might cause trouble if the other UA simply can't handle the 
>>additional m-line. If so it is quite broken.)
>>
>>> comment:
>>> If already-executed changes is made by only PRACK transaction,
>>> IMO UAC need not generate a new offer to synchronize both UAs.
>>
>>I don't understand what point you are making, or why.
>
>As Reconfirmation of fundamental premise:
>
>If reINVITE is rejected, then both UAs must cancel all of
>changes within the reINVITE transaction.
>Originally this rollback is an automatic process, so sending
>new offer is not necessary.
>
>Why should UAC send new offer?
>The reason is that:
>   "There are also certain
>   race condition situations that get both user agents out of
>   synchronization" as shown in Figure 5.
>
>In Figure 5, if UPDATE transaction does not exist, the common
>view of oa state is already synchronized(i.e., sendrecv audio
>and inactive video).
>
>I think in this case UAC need not generate a new offer.
>
>Regards,
>Shinji
>
>>                Thanks,
>>                Paul
>>
>>> Regards,
>>> Shinji
>>> 
>>> Internet-Drafts@ietf.org
>>> Sat,  8 May 2010 00:30:02 -0700 (PDT)
>>>> 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-04.txt
>>>>              Pages           : 24
>>>>              Date            : 2010-05-08
>>>>
>>>> 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.
>>>>
>>>> A URL for this Internet-Draft is:
>>>> http://www.ietf.org/internet-drafts/draft-ietf-sipcore-reinvite-04.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

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From shinji.okumura@softfront.jp  Wed May 12 22:13:30 2010
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From: OKUMURA Shinji <shinji.okumura@softfront.jp>
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Subject: Re: [sipcore] I-D Action:draft-ietf-sipcore-reinvite-04.txt
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Hi,

In this text,

3.5.  Glare Situations
   These rules imply that if the
   UAS of a re-INVITE transaction receives and UPDATE request with an
   offer after having sent a non-2xx final response to the re-INVITE but
   before having received the corresponding ACK request, the UA SHOULD
   return a 491 (Request Pending) response to the UPDATE request.

that is shown in a Figure below.

    A                               B
    |                               |
    |re-INV (offer1)                |
    |------------------------------>|
    |                   18x(answer1)|
    |<------------------------------|
    |                        non-2xx|
    |               /---------------|<-+
    |UPD           /                |  | reject UPDATE
    |=============/================>|* |
    |            /          491(UPD)|  |
    |<==========/===================|  |
    |          /                    |  |
    |<--------/                     |  |
    |ACK                            |  |
    |------------------------------>|<-+
    |                               |

Regarding 491 and 500 response, there are some descriptions in RFC3261
and RFC3311. But in these text, an interworking of INVITE and UPDATE
is not clear.

IMO when the directions of the both request are corresponding,
the response code seems to be 500. And if the directions are not
corresponding, the response code seems to be 491.

Therefore in this case, I think a UAS should return a 500 response to
an UPDATE request.

Regards,
Shinji

Internet-Drafts@ietf.org
Sat,  8 May 2010 00:30:02 -0700 (PDT)
>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-04.txt
>	Pages           : 24
>	Date            : 2010-05-08
>
>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.
>
>A URL for this Internet-Draft is:
>http://www.ietf.org/internet-drafts/draft-ietf-sipcore-reinvite-04.txt
>
>Internet-Drafts are also available by anonymous FTP at:
>ftp://ftp.ietf.org/internet-drafts/
>
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--=_alternative 002133F948257722_=--


From shinji.okumura@softfront.jp  Thu May 13 02:35:43 2010
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From: OKUMURA Shinji <shinji.okumura@softfront.jp>
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Cc: gao.yang2@zte.com.cn
Subject: Re: [sipcore] I-D Action:draft-ietf-sipcore-reinvite-04.txt
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Gao,

IIUC you means that UAS treats a 4xx to INVITE as an offer and
an ACK as an answer.

Well, 

    A                               B
    |                               |
    |re-INV (offer1)                |
    |------------------------------>|
    |                   18x(answer1)|
    |<------------------------------|
    |                        non-2xx|
    |<------------------------------|
    |                               |
    |                   /---------->|*
    |ACK               /            |
    |-----------------/------------>|
    |                /              |
    |re-INV (offer2)/               |
    |--------------/                |
    |                               |

In an above Figure, what is a response code to the 2nd
re-INVITE?
Or if UA B received the 2nd re-INIVTE before non-2xx,
what is a response code to it?

gao.yang2@zte.com.cn
Thu, 13 May 2010 14:00:16 +0800
>Hi Shinji,
>
>There's precondition for the text you mentioned: 
>
>When checking for glare situations, UAs MUST treat the exchange of a
>   non-2xx final response to a re-INVITE and its corresponding ACK
>   request as an offer/answer exchange.  Consequently, the rules
>   regarding glare situations applicable to offer/answer exchanges are
>   also applicable to those exchanges.  These rules imply that if the
>   UAS of a re-INVITE transaction receives and UPDATE request with an
>   offer after having sent a non-2xx final response to the re-INVITE but
>   before having received the corresponding ACK request, the UA SHOULD
>   return a 491 (Request Pending) response to the UPDATE request. 
>
>With the precondition defined here, the UAS(of Re-INVITE) MUST think it 
>has sended a Offer after it send the non-2xx.
>
>Before the UAS get the ACK, it get a new Offer in UPDATE, it SHOULD send 
>4xx.
>
>There's normative definition here.

Is that "normative definition" defined in this draft?

Regards,
Shinji


>Thanks,
>
>Gao
>
>===================================
> Zip    : 210012
> Tel    : 87211
> Tel2   :(+86)-025-52877211
> e_mail : gao.yang2@zte.com.cn
>===================================
>
>OKUMURA Shinji <shinji.okumura@softfront.jp> 
>
>Hi,
>
>In this text,
>
>3.5.  Glare Situations
>   These rules imply that if the
>   UAS of a re-INVITE transaction receives and UPDATE request with an
>   offer after having sent a non-2xx final response to the re-INVITE but
>   before having received the corresponding ACK request, the UA SHOULD
>   return a 491 (Request Pending) response to the UPDATE request.
>
>that is shown in a Figure below.
>
>    A                               B
>    |                               |
>    |re-INV (offer1)                |
>    |------------------------------>|
>    |                   18x(answer1)|
>    |<------------------------------|
>    |                        non-2xx|
>    |               /---------------|<-+
>    |UPD           /                |  | reject UPDATE
>    |=============/================>|* |
>    |            /          491(UPD)|  |
>    |<==========/===================|  |
>    |          /                    |  |
>    |<--------/                     |  |
>    |ACK                            |  |
>    |------------------------------>|<-+
>    |                               |
>
>Regarding 491 and 500 response, there are some descriptions in RFC3261
>and RFC3311. But in these text, an interworking of INVITE and UPDATE
>is not clear.
>
>IMO when the directions of the both request are corresponding,
>the response code seems to be 500. And if the directions are not
>corresponding, the response code seems to be 491.
>
>Therefore in this case, I think a UAS should return a 500 response to
>an UPDATE request.
>
>Regards,
>Shinji
>
>Internet-Drafts@ietf.org
>Sat,  8 May 2010 00:30:02 -0700 (PDT)
>>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-04.txt
>>                Pages           : 24
>>                Date            : 2010-05-08
>>
>>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.
>>
>>A URL for this Internet-Draft is:
>>http://www.ietf.org/internet-drafts/draft-ietf-sipcore-reinvite-04.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.

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From adam@nostrum.com  Thu May 13 08:33:32 2010
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Cc: SIPCORE <sipcore@ietf.org>
Subject: Re: [sipcore] SigComp and Service-Route
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Christer:

Sorry for the slow response. Some responses inline.

On 5/6/10 2:51 PM, Christer Holmberg wrote:
> 1. How can the UA ensure that the outbound proxy will be listed in the
> Service-Route in the first place? RFC 3608 (Service-Route) does not
> mandate the registrar to insert any intermediaries in the Service-Route,
> e.g based on Path. It's all based on local policy.
>    

Service-Route is the means by which the network conveys the proxies that 
the UA needs to use when accessing the network. I think there's an 
assumption in certain networks -- like IMS -- that the registrar will, 
as a matter of policy, copy the Path into the Service-Route.

Strictly speaking, though, even that's not necessary for the UA to 
ensure traversal of a specific first-hop proxy. Service-Route specifies 
a minimal set of proxies to be traversed, not an exhaustive and 
restrictive set of proxies. There is nothing in the Service-Route 
mechanism that prevents the UA from inserting a specific outbound proxy 
in all of its requests.

So, trivially, if the UA discovers that a SigComp-supporting proxy isn't 
the first hop in the Service-Route returned by the network, it makes a 
local decision to send all requests through one that does. (Recall that 
support of SigComp by the first proxy in the Service-Route set can be 
determined by casual inspection: if the URL has a ";comp=sigcomp" 
parameter, it supports SigComp; otherwise, it does not.)

> 2. Even if the registrar would build the Service-Route based on Path,
> how does it know that the Path address/port values will be
> usable/routable in the UA-towards-registrar direction (Path is supposed
> to be used in the registrar-towards-UA direction)? This is similar to
> the Record-Route problem, and intermediaries with different network
> addresses/ports per direction would have to insert "double Path".
>    

If this mechanism is used, and if the outbound proxy presents different 
address spaces towards the UA than it does towards the registrar, then 
the outbound proxy will need to take some extraordinary measures to 
ensure that everything works smoothly. Inserting a "Double-Path" would 
be one approach.

> I guess the outbound proxy could insert itself in the Service-Route, but
> that is not described anywhere in RFC 5049, and it is also a SHOULD NOT
> in RFC 3608.
>    

That approach is congruent to the "change the Record-Route" mechanism 
that 3261 originally described for multihomed proxies. If you'll recall, 
we determined this approach to be problematic enough that we issued a 
correction -- RFC5658 -- that basically says "don't do that."

If you're operating in a network that, as a matter of policy, reflects 
the Path back in a Service-Route, then I think the "Double-Path" 
approach is really what you want to do for multi-homed outbound proxies. 
And, even if you're not, doing a "Double-Path" doesn't cause any issues.

/a

From dean.willis@softarmor.com  Thu May 13 10:55:28 2010
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From: Dean Willis <dean.willis@softarmor.com>
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Cc: "Paul E. Jones" <paulej@packetizer.com>, "sipcore@ietf.org" <sipcore@ietf.org>, Hadriel Kaplan <HKaplan@acmepacket.com>
Subject: Re: [sipcore] The predictive value of OPTIONS
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Paul Kyzivat wrote:

> There is no way to win here. You must be inconsistent at some point. So
> the result of the OPTIONS wrt Accept cannot be trusted to predict future
> behavior. The same is true for most of the other things reported in
> OPTIONS - its really hard to say much about how relevant they are for
> predicting future behavior.

This gets right at the aspect of option-tags that people seem to keep
getting confused on.

OPTIONS responses and option-tags DO NOT assert (or require) anything
about "willingness" or "authorization". They assert only capability.

You can be capable of understanding a MESSAGE request without being
willing to accept one from a given party (or, in the case of a B2BUA, TO
a given party).

We keep falling back into trying to use optionality to predict or
mandate outcome. We MUST NOT do this -- the mechanism just doesn't work
this way. At best, optionality in SIP is a way of predicting
understanding, not behavior.

Consider the difference between Bob approaching Alice and saying "Would
you understand what I was asking if I asked you to go home with me?" vs.
Bob asking Alice "Will you go home with me?"

The first is an optionality discussion; the second is a request for a
specific extension.

--
Dean



--
Dean

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Subject: Re: [sipcore] The predictive value of OPTIONS
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[as a participant]

On 5/13/10 12:54 PM, Dean Willis wrote:
> Consider the difference between Bob approaching Alice and saying "Would
> you understand what I was asking if I asked you to go home with me?" vs.
> Bob asking Alice "Will you go home with me?"
>    

I nominate this for the award of "most illustrative analogy ever applied 
to SIP."

:-)

/a

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On 05/13/2010 01:20 PM, Adam Roach wrote:
> [as a participant]
> 
> On 5/13/10 12:54 PM, Dean Willis wrote:
>> Consider the difference between Bob approaching Alice and saying "Would
>> you understand what I was asking if I asked you to go home with me?" vs.
>> Bob asking Alice "Will you go home with me?"
>>    
> 
> I nominate this for the award of "most illustrative analogy ever applied
> to SIP."

Oddly enough, about 30 seconds ago I sent a message to DISPATCH about
Bob chatting up Alice. Precognition, I guess.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kfleming@digium.com
Check us out at www.digium.com & www.asterisk.org

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Subject: Re: [sipcore] Call for Consensus: INFO Specification and Congestion
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[as a participant]

Regarding the proposed text that I sent to the mailing list last week, 
there was an exchange regarding where in the document this behavior goes;

On 5/2/10 6:57 AM, DRAGE, Keith (Keith) wrote:
> ...[T]his text does not belong in section 10, as section 10 is about what needs to be documented to define an INFO package, not about what the implementor actually needs to do. It actually relates to whether it was appropriate to select INFO in the first place to do the job, as if the information transfer requirements are inappropriate to the underlying transport capabilities of SIP, then INFO should not have been selected in the first place.
>    

On 5/5/10 8:28 AM, Christer Holmberg responded:
> So, I guess it should be somewhere in section 7.4.
>    

On 5/5/10 8:45 AM, DRAGE, Keith (Keith) replied:
> Yes, but obviously not in the same words. It needs to be tailored as do not choose an INFO solution if you need to do this.
>    

I would argue that section 7.3 already goes over this ground in terms of 
whether INFO is appropriate for a particular use case. And, in fact, 
*most* of the proposed text isn't talking about what event package 
designers should do. Some of it does, but for the most part, the text 
talks about what implementors are required to do. The document *does* 
have a section that covers this ground: section 3.

To that end, I would think we could split this into two sections. The 
part dealing with implementation issues would be a new section 3.5; the 
other would form an introduction to the existing section 10.9.

Incorporating the other suggestions made so far, then, I would propose 
the following sections for the INFO document:

> 3.5.  Rate and Size of INFO Requests
>
>      INFO messages differ from many other sorts of SIP messages in
>      that they carry application information, and the size and rate
>      of the INFO message is directly determined by the application.
>      This can cause application information traffic to interfere
>      with other traffic on that infrastructure, or to self-interfere
>      when data rates become too high.
>
>      Implementors need to consider the impact that the size and the
>      rate of the INFO messages have on the network and on other
>      traffic, since it normally cannot be ensured that INFO messages
>      will be carried over a congestion-controlled transport protocol
>      end-to-end.  Even if an INFO message is sent over such a
>      transport protocol, a downstream SIP entity might forward the
>      message over a transport protocol that does not provide
>      congestion control.
>
>      Therefore, an application SHOULD generally have at most one
>      outstanding INFO message.  Implementors are reminded to observe
>      the guidelines in RFC3261 Section 18 regarding the size of SIP
>      messages, such as the guideline to not exceed message sizes
>      of 1300 bytes when the path MTU is unknown. RFC5405 provides
>      additional guidelines for applications using the User Datagram
>      Protocol (UDP) that may be useful background reading.
>
>
>
> 10.9 Rate of INFO Requests
>
>      A designer of an Info package needs to be mindful of the
>      potential congestion impact their application can have on both
>      the SIP network and the Internet at large.  See section 3.5
>      for a discussion of congestion-related issues.
>
>      If there is a maximum or minimum rate at which UAs can send
>      INFO requests associated with the Info Package within a dialog,
>      the Info Package specification MUST document the rate values.
>
>      If the rates can vary, the Info Package specification MAY
>      define Info Package parameters that UAs can use to indicate
>      or negotiate the rates.  Alternatively the rate information
>      can be part of the application data information associated
>      with the Info Package.


Does that sound reasonable?

/a

From keith.drage@alcatel-lucent.com  Sun May 16 15:03:53 2010
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Subject: Re: [sipcore] Call for Consensus: INFO Specification and Congestion
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This proposed paragraph still causes me issues:

>      Therefore, an application SHOULD generally have at most one
>      outstanding INFO message.  Implementors are reminded to observe
>      the guidelines in RFC3261 Section 18 regarding the size of SIP
>      messages, such as the guideline to not exceed message sizes
>      of 1300 bytes when the path MTU is unknown. RFC5405 provides
>      additional guidelines for applications using the User Datagram
>      Protocol (UDP) that may be useful background reading.

I already objected to the first sentence above as I do not believe it solve=
s the issue, and removes useful capabilities that may be desired by some ap=
plications. I CANNOT accept this sentence.

The second sentence above is an inexact extraction, and as such misleads co=
mpletely. The correct quote is below:

   If a request is within 200 bytes of the path MTU, or if it is larger
   than 1300 bytes and the path MTU is unknown, the request MUST be sent
   using an RFC 2914 [43] congestion controlled transport protocol, such
   as TCP. If this causes a change in the transport protocol from the
   one indicated in the top Via, the value in the top Via MUST be
   changed.  This prevents fragmentation of messages over UDP and
   provides congestion control for larger messages.  However,
   implementations MUST be able to handle messages up to the maximum
   datagram packet size.  For UDP, this size is 65,535 bytes, including
   IP and UDP headers.

Nothing about not exceeding message sizes. Only restriction is to send usin=
g a congestion controlled transport protocol if it exceeds MTU size. Theref=
ore second sentence as currently drafted is unacceptable.


regards

Keith
=20

> -----Original Message-----
> From: Adam Roach [mailto:adam@nostrum.com]=20
> Sent: Thursday, May 13, 2010 8:12 PM
> To: DRAGE, Keith (Keith)
> Cc: Christer Holmberg; Adam Roach; SIPCORE; lars.eggert@nokia.com
> Subject: Re: [sipcore] Call for Consensus: INFO Specification=20
> and Congestion
>=20
> [as a participant]
>=20
> Regarding the proposed text that I sent to the mailing list=20
> last week, there was an exchange regarding where in the=20
> document this behavior goes;
>=20
> On 5/2/10 6:57 AM, DRAGE, Keith (Keith) wrote:
> > ...[T]his text does not belong in section 10, as section 10=20
> is about what needs to be documented to define an INFO=20
> package, not about what the implementor actually needs to do.=20
> It actually relates to whether it was appropriate to select=20
> INFO in the first place to do the job, as if the information=20
> transfer requirements are inappropriate to the underlying=20
> transport capabilities of SIP, then INFO should not have been=20
> selected in the first place.
> >   =20
>=20
> On 5/5/10 8:28 AM, Christer Holmberg responded:
> > So, I guess it should be somewhere in section 7.4.
> >   =20
>=20
> On 5/5/10 8:45 AM, DRAGE, Keith (Keith) replied:
> > Yes, but obviously not in the same words. It needs to be=20
> tailored as do not choose an INFO solution if you need to do this.
> >   =20
>=20
> I would argue that section 7.3 already goes over this ground=20
> in terms of whether INFO is appropriate for a particular use=20
> case. And, in fact,
> *most* of the proposed text isn't talking about what event=20
> package designers should do. Some of it does, but for the=20
> most part, the text talks about what implementors are=20
> required to do. The document *does* have a section that=20
> covers this ground: section 3.
>=20
> To that end, I would think we could split this into two=20
> sections. The part dealing with implementation issues would=20
> be a new section 3.5; the other would form an introduction to=20
> the existing section 10.9.
>=20
> Incorporating the other suggestions made so far, then, I=20
> would propose the following sections for the INFO document:
>=20
> > 3.5.  Rate and Size of INFO Requests
> >
> >      INFO messages differ from many other sorts of SIP messages in
> >      that they carry application information, and the size and rate
> >      of the INFO message is directly determined by the application.
> >      This can cause application information traffic to interfere
> >      with other traffic on that infrastructure, or to self-interfere
> >      when data rates become too high.
> >
> >      Implementors need to consider the impact that the size and the
> >      rate of the INFO messages have on the network and on other
> >      traffic, since it normally cannot be ensured that INFO messages
> >      will be carried over a congestion-controlled transport protocol
> >      end-to-end.  Even if an INFO message is sent over such a
> >      transport protocol, a downstream SIP entity might forward the
> >      message over a transport protocol that does not provide
> >      congestion control.
> >
> >      Therefore, an application SHOULD generally have at most one
> >      outstanding INFO message.  Implementors are reminded to observe
> >      the guidelines in RFC3261 Section 18 regarding the size of SIP
> >      messages, such as the guideline to not exceed message sizes
> >      of 1300 bytes when the path MTU is unknown. RFC5405 provides
> >      additional guidelines for applications using the User Datagram
> >      Protocol (UDP) that may be useful background reading.
> >
> >
> >
> > 10.9 Rate of INFO Requests
> >
> >      A designer of an Info package needs to be mindful of the
> >      potential congestion impact their application can have on both
> >      the SIP network and the Internet at large.  See section 3.5
> >      for a discussion of congestion-related issues.
> >
> >      If there is a maximum or minimum rate at which UAs can send
> >      INFO requests associated with the Info Package within a dialog,
> >      the Info Package specification MUST document the rate values.
> >
> >      If the rates can vary, the Info Package specification MAY
> >      define Info Package parameters that UAs can use to indicate
> >      or negotiate the rates.  Alternatively the rate information
> >      can be part of the application data information associated
> >      with the Info Package.
>=20
>=20
> Does that sound reasonable?
>=20
> /a
> =

From christer.holmberg@ericsson.com  Sun May 16 22:35:21 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>, Adam Roach <adam@nostrum.com>
Date: Mon, 17 May 2010 07:31:48 +0200
Thread-Topic: [sipcore] Call for Consensus: INFO Specification and Congestion
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Cc: Adam Roach <adam@estacado.net>, SIPCORE <sipcore@ietf.org>, "lars.eggert@nokia.com" <lars.eggert@nokia.com>
Subject: Re: [sipcore] Call for Consensus: INFO Specification and Congestion
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Hi,

I appologize that it has taken me a while to reply on this issue. I've been=
 busy with 3GPP the last few weeks.


Note that the draft already contains section 7.3, which talks about rate an=
d volume. Nobody has commented on that, so I assume it is ok.

My suggestion would be to:

-	Remove section 7.3
-	Add new section 7.4.1.2, that talks about rate and volume
-	New section 7.4.1.2 consists of text from old section 7.3 and the suggest=
ed new text for section 10.9.
-	Keep section 10.9 unchanged (draft version -07)

So, new section 7.4.1.2 could look like:


"7.4.1.2.  INFO Request Rate and Volume

   INFO messages differ from many other sorts of SIP messages in that
   they carry application information, and the size and rate of the
   INFO message is directly determined by the application. This can
   cause application information traffic to interfere with other traffic
   on that infrastructure, or to self-interfere when data rates become
   too high.

   There is no default throttling mechanism for INFO requests. Apart
   from the SIP session establishment, the number of SIP messages
   exchanged during the lifetime a normal SIP session is rather small.

   Some applications, like sending of DTMF tones, can generate a burst
   of up to 20 messages per second.  Other applications, like constant
   GPS location updates, could generate a high rate of INFO requests
   during the lifetime of the invite dialog usage.

   A designer of an INFO package, and the application that uses it,
   need to consider the impact that the size and the rate of the INFO
   messages have on the network and on other traffic, since it normally
   cannot be ensured that INFO messages will be carried over a
   congestion-controlled transport protocol end-to-end. Even if an
   INFO message is sent over a such a transport protocol, a downstream
   SIP entity might forward the message over a transport protocol that
   does not provide congestion control.

   Furthermore, SIP messages tend to be relatively small, on the order
   of 500 Bytes to 32K Bytes. SIP is a poor mechanism for direct
   exchange of bulk data beyond these limits, especially if the headers
   plus body exceed the UDP MTU [RFC0768]. Appropriate mechanisms for
   such traffic include HTTP [RFC2616], MSRP [RFC4975], or other media
   plane data transport mechanisms.

   RFC5405 provides additional guidelines for applications using the=20
   User Datagram Protocol (UDP) that may be useful background reading."

And, section 10.9 would look like:

"10.9.  Rate of INFO Requests

   If there is a maximum or minimum rate at which UAs can send INFO
   requests associated with the Info Package within a dialog, the Info
   Package specification MUST document the rate values.

   If the rates can vary, the Info Package specification MAY define Info
   Package parameters that UAs can use to indicate or negotiate the
   rates.  Alternatively the rate information can be part of the
   application data information associated with the Info Package."


Would people be ok with that?

Regards,

Christer


























=20

> -----Original Message-----
> From: DRAGE, Keith (Keith) [mailto:keith.drage@alcatel-lucent.com]=20
> Sent: 17. toukokuuta 2010 1:03
> To: Adam Roach
> Cc: Christer Holmberg; Adam Roach; SIPCORE; lars.eggert@nokia.com
> Subject: RE: [sipcore] Call for Consensus: INFO Specification=20
> and Congestion
>=20
> This proposed paragraph still causes me issues:
>=20
> >      Therefore, an application SHOULD generally have at most one
> >      outstanding INFO message.  Implementors are reminded to observe
> >      the guidelines in RFC3261 Section 18 regarding the size of SIP
> >      messages, such as the guideline to not exceed message sizes
> >      of 1300 bytes when the path MTU is unknown. RFC5405 provides
> >      additional guidelines for applications using the User Datagram
> >      Protocol (UDP) that may be useful background reading.
>=20
> I already objected to the first sentence above as I do not=20
> believe it solves the issue, and removes useful capabilities=20
> that may be desired by some applications. I CANNOT accept=20
> this sentence.
>=20
> The second sentence above is an inexact extraction, and as=20
> such misleads completely. The correct quote is below:
>=20
>    If a request is within 200 bytes of the path MTU, or if it=20
> is larger
>    than 1300 bytes and the path MTU is unknown, the request=20
> MUST be sent
>    using an RFC 2914 [43] congestion controlled transport=20
> protocol, such
>    as TCP. If this causes a change in the transport protocol from the
>    one indicated in the top Via, the value in the top Via MUST be
>    changed.  This prevents fragmentation of messages over UDP and
>    provides congestion control for larger messages.  However,
>    implementations MUST be able to handle messages up to the maximum
>    datagram packet size.  For UDP, this size is 65,535 bytes,=20
> including
>    IP and UDP headers.
>=20
> Nothing about not exceeding message sizes. Only restriction=20
> is to send using a congestion controlled transport protocol=20
> if it exceeds MTU size. Therefore second sentence as=20
> currently drafted is unacceptable.
>=20
>=20
> regards
>=20
> Keith
> =20
>=20
> > -----Original Message-----
> > From: Adam Roach [mailto:adam@nostrum.com]
> > Sent: Thursday, May 13, 2010 8:12 PM
> > To: DRAGE, Keith (Keith)
> > Cc: Christer Holmberg; Adam Roach; SIPCORE; lars.eggert@nokia.com
> > Subject: Re: [sipcore] Call for Consensus: INFO Specification and=20
> > Congestion
> >=20
> > [as a participant]
> >=20
> > Regarding the proposed text that I sent to the mailing list=20
> last week,=20
> > there was an exchange regarding where in the document this behavior=20
> > goes;
> >=20
> > On 5/2/10 6:57 AM, DRAGE, Keith (Keith) wrote:
> > > ...[T]his text does not belong in section 10, as section 10
> > is about what needs to be documented to define an INFO package, not=20
> > about what the implementor actually needs to do.
> > It actually relates to whether it was appropriate to select INFO in=20
> > the first place to do the job, as if the information transfer=20
> > requirements are inappropriate to the underlying transport=20
> > capabilities of SIP, then INFO should not have been selected in the=20
> > first place.
> > >   =20
> >=20
> > On 5/5/10 8:28 AM, Christer Holmberg responded:
> > > So, I guess it should be somewhere in section 7.4.
> > >   =20
> >=20
> > On 5/5/10 8:45 AM, DRAGE, Keith (Keith) replied:
> > > Yes, but obviously not in the same words. It needs to be
> > tailored as do not choose an INFO solution if you need to do this.
> > >   =20
> >=20
> > I would argue that section 7.3 already goes over this=20
> ground in terms=20
> > of whether INFO is appropriate for a particular use case. And, in=20
> > fact,
> > *most* of the proposed text isn't talking about what event package=20
> > designers should do. Some of it does, but for the most=20
> part, the text=20
> > talks about what implementors are required to do. The=20
> document *does*=20
> > have a section that covers this ground: section 3.
> >=20
> > To that end, I would think we could split this into two=20
> sections. The=20
> > part dealing with implementation issues would be a new section 3.5;=20
> > the other would form an introduction to the existing section 10.9.
> >=20
> > Incorporating the other suggestions made so far, then, I=20
> would propose=20
> > the following sections for the INFO document:
> >=20
> > > 3.5.  Rate and Size of INFO Requests
> > >
> > >      INFO messages differ from many other sorts of SIP messages in
> > >      that they carry application information, and the=20
> size and rate
> > >      of the INFO message is directly determined by the=20
> application.
> > >      This can cause application information traffic to interfere
> > >      with other traffic on that infrastructure, or to=20
> self-interfere
> > >      when data rates become too high.
> > >
> > >      Implementors need to consider the impact that the=20
> size and the
> > >      rate of the INFO messages have on the network and on other
> > >      traffic, since it normally cannot be ensured that=20
> INFO messages
> > >      will be carried over a congestion-controlled=20
> transport protocol
> > >      end-to-end.  Even if an INFO message is sent over such a
> > >      transport protocol, a downstream SIP entity might forward the
> > >      message over a transport protocol that does not provide
> > >      congestion control.
> > >
> > >      Therefore, an application SHOULD generally have at most one
> > >      outstanding INFO message.  Implementors are reminded=20
> to observe
> > >      the guidelines in RFC3261 Section 18 regarding the=20
> size of SIP
> > >      messages, such as the guideline to not exceed message sizes
> > >      of 1300 bytes when the path MTU is unknown. RFC5405 provides
> > >      additional guidelines for applications using the=20
> User Datagram
> > >      Protocol (UDP) that may be useful background reading.
> > >
> > >
> > >
> > > 10.9 Rate of INFO Requests
> > >
> > >      A designer of an Info package needs to be mindful of the
> > >      potential congestion impact their application can=20
> have on both
> > >      the SIP network and the Internet at large.  See section 3.5
> > >      for a discussion of congestion-related issues.
> > >
> > >      If there is a maximum or minimum rate at which UAs can send
> > >      INFO requests associated with the Info Package=20
> within a dialog,
> > >      the Info Package specification MUST document the rate values.
> > >
> > >      If the rates can vary, the Info Package specification MAY
> > >      define Info Package parameters that UAs can use to indicate
> > >      or negotiate the rates.  Alternatively the rate information
> > >      can be part of the application data information associated
> > >      with the Info Package.
> >=20
> >=20
> > Does that sound reasonable?
> >=20
> > /a
> > =

From keith.drage@alcatel-lucent.com  Mon May 17 02:30:58 2010
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Adam Roach <adam@nostrum.com>
Date: Mon, 17 May 2010 11:27:53 +0200
Thread-Topic: [sipcore] Call for Consensus: INFO Specification and Congestion
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Subject: Re: [sipcore] Call for Consensus: INFO Specification and Congestion
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I'm OK with this proposal.

Keith=20

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: Monday, May 17, 2010 6:32 AM
> To: DRAGE, Keith (Keith); Adam Roach
> Cc: Adam Roach; SIPCORE; lars.eggert@nokia.com
> Subject: RE: [sipcore] Call for Consensus: INFO Specification=20
> and Congestion
>=20
>=20
> Hi,
>=20
> I appologize that it has taken me a while to reply on this=20
> issue. I've been busy with 3GPP the last few weeks.
>=20
>=20
> Note that the draft already contains section 7.3, which talks=20
> about rate and volume. Nobody has commented on that, so I=20
> assume it is ok.
>=20
> My suggestion would be to:
>=20
> -	Remove section 7.3
> -	Add new section 7.4.1.2, that talks about rate and volume
> -	New section 7.4.1.2 consists of text from old section=20
> 7.3 and the suggested new text for section 10.9.
> -	Keep section 10.9 unchanged (draft version -07)
>=20
> So, new section 7.4.1.2 could look like:
>=20
>=20
> "7.4.1.2.  INFO Request Rate and Volume
>=20
>    INFO messages differ from many other sorts of SIP messages in that
>    they carry application information, and the size and rate of the
>    INFO message is directly determined by the application. This can
>    cause application information traffic to interfere with=20
> other traffic
>    on that infrastructure, or to self-interfere when data rates become
>    too high.
>=20
>    There is no default throttling mechanism for INFO requests. Apart
>    from the SIP session establishment, the number of SIP messages
>    exchanged during the lifetime a normal SIP session is rather small.
>=20
>    Some applications, like sending of DTMF tones, can generate a burst
>    of up to 20 messages per second.  Other applications, like constant
>    GPS location updates, could generate a high rate of INFO requests
>    during the lifetime of the invite dialog usage.
>=20
>    A designer of an INFO package, and the application that uses it,
>    need to consider the impact that the size and the rate of the INFO
>    messages have on the network and on other traffic, since=20
> it normally
>    cannot be ensured that INFO messages will be carried over a
>    congestion-controlled transport protocol end-to-end. Even if an
>    INFO message is sent over a such a transport protocol, a downstream
>    SIP entity might forward the message over a transport protocol that
>    does not provide congestion control.
>=20
>    Furthermore, SIP messages tend to be relatively small, on the order
>    of 500 Bytes to 32K Bytes. SIP is a poor mechanism for direct
>    exchange of bulk data beyond these limits, especially if=20
> the headers
>    plus body exceed the UDP MTU [RFC0768]. Appropriate mechanisms for
>    such traffic include HTTP [RFC2616], MSRP [RFC4975], or other media
>    plane data transport mechanisms.
>=20
>    RFC5405 provides additional guidelines for applications using the=20
>    User Datagram Protocol (UDP) that may be useful background=20
> reading."
>=20
> And, section 10.9 would look like:
>=20
> "10.9.  Rate of INFO Requests
>=20
>    If there is a maximum or minimum rate at which UAs can send INFO
>    requests associated with the Info Package within a dialog, the Info
>    Package specification MUST document the rate values.
>=20
>    If the rates can vary, the Info Package specification MAY=20
> define Info
>    Package parameters that UAs can use to indicate or negotiate the
>    rates.  Alternatively the rate information can be part of the
>    application data information associated with the Info Package."
>=20
>=20
> Would people be ok with that?
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>=20
>=20
> =20
>=20
> > -----Original Message-----
> > From: DRAGE, Keith (Keith) [mailto:keith.drage@alcatel-lucent.com]
> > Sent: 17. toukokuuta 2010 1:03
> > To: Adam Roach
> > Cc: Christer Holmberg; Adam Roach; SIPCORE; lars.eggert@nokia.com
> > Subject: RE: [sipcore] Call for Consensus: INFO Specification and=20
> > Congestion
> >=20
> > This proposed paragraph still causes me issues:
> >=20
> > >      Therefore, an application SHOULD generally have at most one
> > >      outstanding INFO message.  Implementors are reminded=20
> to observe
> > >      the guidelines in RFC3261 Section 18 regarding the=20
> size of SIP
> > >      messages, such as the guideline to not exceed message sizes
> > >      of 1300 bytes when the path MTU is unknown. RFC5405 provides
> > >      additional guidelines for applications using the=20
> User Datagram
> > >      Protocol (UDP) that may be useful background reading.
> >=20
> > I already objected to the first sentence above as I do not=20
> believe it=20
> > solves the issue, and removes useful capabilities that may=20
> be desired=20
> > by some applications. I CANNOT accept this sentence.
> >=20
> > The second sentence above is an inexact extraction, and as such=20
> > misleads completely. The correct quote is below:
> >=20
> >    If a request is within 200 bytes of the path MTU, or if it is=20
> > larger
> >    than 1300 bytes and the path MTU is unknown, the request MUST be=20
> > sent
> >    using an RFC 2914 [43] congestion controlled transport protocol,=20
> > such
> >    as TCP. If this causes a change in the transport=20
> protocol from the
> >    one indicated in the top Via, the value in the top Via MUST be
> >    changed.  This prevents fragmentation of messages over UDP and
> >    provides congestion control for larger messages.  However,
> >    implementations MUST be able to handle messages up to the maximum
> >    datagram packet size.  For UDP, this size is 65,535 bytes,=20
> > including
> >    IP and UDP headers.
> >=20
> > Nothing about not exceeding message sizes. Only restriction=20
> is to send=20
> > using a congestion controlled transport protocol if it exceeds MTU=20
> > size. Therefore second sentence as currently drafted is=20
> unacceptable.
> >=20
> >=20
> > regards
> >=20
> > Keith
> > =20
> >=20
> > > -----Original Message-----
> > > From: Adam Roach [mailto:adam@nostrum.com]
> > > Sent: Thursday, May 13, 2010 8:12 PM
> > > To: DRAGE, Keith (Keith)
> > > Cc: Christer Holmberg; Adam Roach; SIPCORE; lars.eggert@nokia.com
> > > Subject: Re: [sipcore] Call for Consensus: INFO Specification and=20
> > > Congestion
> > >=20
> > > [as a participant]
> > >=20
> > > Regarding the proposed text that I sent to the mailing list=20
> > last week,=20
> > > there was an exchange regarding where in the document=20
> this behavior=20
> > > goes;
> > >=20
> > > On 5/2/10 6:57 AM, DRAGE, Keith (Keith) wrote:
> > > > ...[T]his text does not belong in section 10, as section 10
> > > is about what needs to be documented to define an INFO=20
> package, not=20
> > > about what the implementor actually needs to do.
> > > It actually relates to whether it was appropriate to=20
> select INFO in=20
> > > the first place to do the job, as if the information transfer=20
> > > requirements are inappropriate to the underlying transport=20
> > > capabilities of SIP, then INFO should not have been=20
> selected in the=20
> > > first place.
> > > >   =20
> > >=20
> > > On 5/5/10 8:28 AM, Christer Holmberg responded:
> > > > So, I guess it should be somewhere in section 7.4.
> > > >   =20
> > >=20
> > > On 5/5/10 8:45 AM, DRAGE, Keith (Keith) replied:
> > > > Yes, but obviously not in the same words. It needs to be
> > > tailored as do not choose an INFO solution if you need to do this.
> > > >   =20
> > >=20
> > > I would argue that section 7.3 already goes over this=20
> > ground in terms=20
> > > of whether INFO is appropriate for a particular use case. And, in=20
> > > fact,
> > > *most* of the proposed text isn't talking about what=20
> event package=20
> > > designers should do. Some of it does, but for the most=20
> > part, the text=20
> > > talks about what implementors are required to do. The=20
> > document *does*=20
> > > have a section that covers this ground: section 3.
> > >=20
> > > To that end, I would think we could split this into two=20
> > sections. The=20
> > > part dealing with implementation issues would be a new=20
> section 3.5;=20
> > > the other would form an introduction to the existing section 10.9.
> > >=20
> > > Incorporating the other suggestions made so far, then, I=20
> > would propose=20
> > > the following sections for the INFO document:
> > >=20
> > > > 3.5.  Rate and Size of INFO Requests
> > > >
> > > >      INFO messages differ from many other sorts of SIP=20
> messages in
> > > >      that they carry application information, and the=20
> > size and rate
> > > >      of the INFO message is directly determined by the=20
> > application.
> > > >      This can cause application information traffic to interfere
> > > >      with other traffic on that infrastructure, or to=20
> > self-interfere
> > > >      when data rates become too high.
> > > >
> > > >      Implementors need to consider the impact that the=20
> > size and the
> > > >      rate of the INFO messages have on the network and on other
> > > >      traffic, since it normally cannot be ensured that=20
> > INFO messages
> > > >      will be carried over a congestion-controlled=20
> > transport protocol
> > > >      end-to-end.  Even if an INFO message is sent over such a
> > > >      transport protocol, a downstream SIP entity might=20
> forward the
> > > >      message over a transport protocol that does not provide
> > > >      congestion control.
> > > >
> > > >      Therefore, an application SHOULD generally have at most one
> > > >      outstanding INFO message.  Implementors are reminded=20
> > to observe
> > > >      the guidelines in RFC3261 Section 18 regarding the=20
> > size of SIP
> > > >      messages, such as the guideline to not exceed message sizes
> > > >      of 1300 bytes when the path MTU is unknown.=20
> RFC5405 provides
> > > >      additional guidelines for applications using the=20
> > User Datagram
> > > >      Protocol (UDP) that may be useful background reading.
> > > >
> > > >
> > > >
> > > > 10.9 Rate of INFO Requests
> > > >
> > > >      A designer of an Info package needs to be mindful of the
> > > >      potential congestion impact their application can=20
> > have on both
> > > >      the SIP network and the Internet at large.  See section 3.5
> > > >      for a discussion of congestion-related issues.
> > > >
> > > >      If there is a maximum or minimum rate at which UAs can send
> > > >      INFO requests associated with the Info Package=20
> > within a dialog,
> > > >      the Info Package specification MUST document the=20
> rate values.
> > > >
> > > >      If the rates can vary, the Info Package specification MAY
> > > >      define Info Package parameters that UAs can use to indicate
> > > >      or negotiate the rates.  Alternatively the rate information
> > > >      can be part of the application data information associated
> > > >      with the Info Package.
> > >=20
> > >=20
> > > Does that sound reasonable?
> > >=20
> > > /a
> > > =

From christer.holmberg@ericsson.com  Mon May 17 05:50:03 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adam Roach <adam@nostrum.com>
Date: Mon, 17 May 2010 14:46:54 +0200
Thread-Topic: [sipcore] SigComp and Service-Route
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Subject: Re: [sipcore] SigComp and Service-Route
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Hi Adam,=20

>Sorry for the slow response. Some responses inline.
>=20
>On 5/6/10 2:51 PM, Christer Holmberg wrote:
>>1. How can the UA ensure that the outbound proxy will be=20
>>listed in the Service-Route in the first place? RFC 3608 (Service-Route) =
does not=20
>>mandate the registrar to insert any intermediaries in the=20
>>Service-Route, e.g based on Path. It's all based on local policy.   =20
>=20
>Service-Route is the means by which the network conveys the=20
>proxies that the UA needs to use when accessing the network.=20
>I think there's an assumption in certain networks -- like IMS=20
>-- that the registrar will, as a matter of policy, copy the=20
>Path into the Service-Route.

That is not correct - at least not for IMS.

Especially in roaming scenarios, where the P-CSCF (outbound proxy) and S-CS=
CF (registrar) are in different networks, no such assumption can be made.

=20
>Strictly speaking, though, even that's not necessary for the=20
>UA to ensure traversal of a specific first-hop proxy.=20
>Service-Route specifies a minimal set of proxies to be=20
>traversed, not an exhaustive and restrictive set of proxies.=20
>There is nothing in the Service-Route mechanism that prevents=20
>the UA from inserting a specific outbound proxy in all of its=20
>requests.

Correct. An IMS UE for example inserts a P-CSCF (the network may also inser=
t IBCFs) in its requests no matter what the Service-Route contains.

But, the problem is that the UE would need to get a Service-Route including=
 the P-CSCF, in order to get the "comp" parameter.


>So, trivially, if the UA discovers that a SigComp-supporting=20
>proxy isn't the first hop in the Service-Route returned by=20
>the network, it makes a local decision to send all requests=20
>through one that does. (Recall that support of SigComp by the=20
>first proxy in the Service-Route set can be determined by=20
>casual inspection: if the URL has a ";comp=3Dsigcomp"=20
>parameter, it supports SigComp; otherwise, it does not.)

Correct. The problem is that so far the P-CSCF is unable to get ";comp=3Dsi=
gcomp" inserted into the Service-Route, since the P-CSCF cannot be sure tha=
t the S-CSCF will insert it.


>>2. Even if the registrar would build the Service-Route based on Path,=20
>>how does it know that the Path address/port values will be=20
>>usable/routable in the UA-towards-registrar direction (Path is=20
>>supposed to be used in the registrar-towards-UA direction)? This is=20
>>similar to the Record-Route problem, and intermediaries=20
>>with different network addresses/ports per direction would have to insert=
=20
>>"double Path".
>>   =20
>=20
>If this mechanism is used, and if the outbound proxy presents=20
>different address spaces towards the UA than it does towards=20
>the registrar, then the outbound proxy will need to take some=20
>extraordinary measures to ensure that everything works=20
>smoothly. Inserting a "Double-Path" would be one approach.
>
>>I guess the outbound proxy could insert itself in the Service-Route,=20
>>but that is not described anywhere in RFC 5049, and it is also a=20
>>SHOULD NOT in RFC 3608.
>>   =20
>=20
>That approach is congruent to the "change the Record-Route"=20
>mechanism that 3261 originally described for multihomed=20
>proxies. If you'll recall, we determined this approach to be=20
>problematic enough that we issued a correction -- RFC5658 --=20
>that basically says "don't do that."
>=20
>If you're operating in a network that, as a matter of policy,=20
>reflects the Path back in a Service-Route, then I think the=20
>"Double-Path" approach is really what you want to do for multi-homed=20
>outbound proxies.=20
>And, even if you're not, doing a "Double-Path" doesn't cause=20
>any issues.

Yes. The problem is that the registrar would need to know that the P-CSCF (=
and and any other intermediate inserting Path) has used "Double-Path", in o=
rder to "safely" building the Service-Route based on the path. It may be po=
ssible to configure within a single network, but more difficult in roaming =
scenarios.

Some years ago, Jonathan worked on a draft which was related to this:

http://tools.ietf.org/id/draft-rosenberg-sip-route-construct-02.txt

He did not have time to continue, and said I could take it over if I want t=
o, but I never did it either.

Regards,

Christer



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To: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
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Cc: Adam Roach <adam@estacado.net>, SIPCORE <sipcore@ietf.org>, "lars.eggert@nokia.com" <lars.eggert@nokia.com>
Subject: Re: [sipcore] Call for Consensus: INFO Specification and Congestion
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Maybe this is moot because Keith has agreed to the latest proposal...

I was wondering about your objection to recommending at most one 
outstanding INFO message - because there are useful things to do with this.

In general, having multiple outstanding requests in a dialog is 
problematic. With retransmission of lost messages this can result in 
some messages being rejected due to CSeq errors. While this isn't 
necessarily fatal, and there can be a few cases where this can still 
have utility, those cases are rare.

	Thanks,
	Paul

DRAGE, Keith (Keith) wrote:
> I'm OK with this proposal.
> 
> Keith 
> 
>> -----Original Message-----
>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com] 
>> Sent: Monday, May 17, 2010 6:32 AM
>> To: DRAGE, Keith (Keith); Adam Roach
>> Cc: Adam Roach; SIPCORE; lars.eggert@nokia.com
>> Subject: RE: [sipcore] Call for Consensus: INFO Specification 
>> and Congestion
>>
>>
>> Hi,
>>
>> I appologize that it has taken me a while to reply on this 
>> issue. I've been busy with 3GPP the last few weeks.
>>
>>
>> Note that the draft already contains section 7.3, which talks 
>> about rate and volume. Nobody has commented on that, so I 
>> assume it is ok.
>>
>> My suggestion would be to:
>>
>> -	Remove section 7.3
>> -	Add new section 7.4.1.2, that talks about rate and volume
>> -	New section 7.4.1.2 consists of text from old section 
>> 7.3 and the suggested new text for section 10.9.
>> -	Keep section 10.9 unchanged (draft version -07)
>>
>> So, new section 7.4.1.2 could look like:
>>
>>
>> "7.4.1.2.  INFO Request Rate and Volume
>>
>>    INFO messages differ from many other sorts of SIP messages in that
>>    they carry application information, and the size and rate of the
>>    INFO message is directly determined by the application. This can
>>    cause application information traffic to interfere with 
>> other traffic
>>    on that infrastructure, or to self-interfere when data rates become
>>    too high.
>>
>>    There is no default throttling mechanism for INFO requests. Apart
>>    from the SIP session establishment, the number of SIP messages
>>    exchanged during the lifetime a normal SIP session is rather small.
>>
>>    Some applications, like sending of DTMF tones, can generate a burst
>>    of up to 20 messages per second.  Other applications, like constant
>>    GPS location updates, could generate a high rate of INFO requests
>>    during the lifetime of the invite dialog usage.
>>
>>    A designer of an INFO package, and the application that uses it,
>>    need to consider the impact that the size and the rate of the INFO
>>    messages have on the network and on other traffic, since 
>> it normally
>>    cannot be ensured that INFO messages will be carried over a
>>    congestion-controlled transport protocol end-to-end. Even if an
>>    INFO message is sent over a such a transport protocol, a downstream
>>    SIP entity might forward the message over a transport protocol that
>>    does not provide congestion control.
>>
>>    Furthermore, SIP messages tend to be relatively small, on the order
>>    of 500 Bytes to 32K Bytes. SIP is a poor mechanism for direct
>>    exchange of bulk data beyond these limits, especially if 
>> the headers
>>    plus body exceed the UDP MTU [RFC0768]. Appropriate mechanisms for
>>    such traffic include HTTP [RFC2616], MSRP [RFC4975], or other media
>>    plane data transport mechanisms.
>>
>>    RFC5405 provides additional guidelines for applications using the 
>>    User Datagram Protocol (UDP) that may be useful background 
>> reading."
>>
>> And, section 10.9 would look like:
>>
>> "10.9.  Rate of INFO Requests
>>
>>    If there is a maximum or minimum rate at which UAs can send INFO
>>    requests associated with the Info Package within a dialog, the Info
>>    Package specification MUST document the rate values.
>>
>>    If the rates can vary, the Info Package specification MAY 
>> define Info
>>    Package parameters that UAs can use to indicate or negotiate the
>>    rates.  Alternatively the rate information can be part of the
>>    application data information associated with the Info Package."
>>
>>
>> Would people be ok with that?
>>
>> Regards,
>>
>> Christer
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>  
>>
>>> -----Original Message-----
>>> From: DRAGE, Keith (Keith) [mailto:keith.drage@alcatel-lucent.com]
>>> Sent: 17. toukokuuta 2010 1:03
>>> To: Adam Roach
>>> Cc: Christer Holmberg; Adam Roach; SIPCORE; lars.eggert@nokia.com
>>> Subject: RE: [sipcore] Call for Consensus: INFO Specification and 
>>> Congestion
>>>
>>> This proposed paragraph still causes me issues:
>>>
>>>>      Therefore, an application SHOULD generally have at most one
>>>>      outstanding INFO message.  Implementors are reminded 
>> to observe
>>>>      the guidelines in RFC3261 Section 18 regarding the 
>> size of SIP
>>>>      messages, such as the guideline to not exceed message sizes
>>>>      of 1300 bytes when the path MTU is unknown. RFC5405 provides
>>>>      additional guidelines for applications using the 
>> User Datagram
>>>>      Protocol (UDP) that may be useful background reading.
>>> I already objected to the first sentence above as I do not 
>> believe it 
>>> solves the issue, and removes useful capabilities that may 
>> be desired 
>>> by some applications. I CANNOT accept this sentence.
>>>
>>> The second sentence above is an inexact extraction, and as such 
>>> misleads completely. The correct quote is below:
>>>
>>>    If a request is within 200 bytes of the path MTU, or if it is 
>>> larger
>>>    than 1300 bytes and the path MTU is unknown, the request MUST be 
>>> sent
>>>    using an RFC 2914 [43] congestion controlled transport protocol, 
>>> such
>>>    as TCP. If this causes a change in the transport 
>> protocol from the
>>>    one indicated in the top Via, the value in the top Via MUST be
>>>    changed.  This prevents fragmentation of messages over UDP and
>>>    provides congestion control for larger messages.  However,
>>>    implementations MUST be able to handle messages up to the maximum
>>>    datagram packet size.  For UDP, this size is 65,535 bytes, 
>>> including
>>>    IP and UDP headers.
>>>
>>> Nothing about not exceeding message sizes. Only restriction 
>> is to send 
>>> using a congestion controlled transport protocol if it exceeds MTU 
>>> size. Therefore second sentence as currently drafted is 
>> unacceptable.
>>>
>>> regards
>>>
>>> Keith
>>>  
>>>
>>>> -----Original Message-----
>>>> From: Adam Roach [mailto:adam@nostrum.com]
>>>> Sent: Thursday, May 13, 2010 8:12 PM
>>>> To: DRAGE, Keith (Keith)
>>>> Cc: Christer Holmberg; Adam Roach; SIPCORE; lars.eggert@nokia.com
>>>> Subject: Re: [sipcore] Call for Consensus: INFO Specification and 
>>>> Congestion
>>>>
>>>> [as a participant]
>>>>
>>>> Regarding the proposed text that I sent to the mailing list 
>>> last week, 
>>>> there was an exchange regarding where in the document 
>> this behavior 
>>>> goes;
>>>>
>>>> On 5/2/10 6:57 AM, DRAGE, Keith (Keith) wrote:
>>>>> ...[T]his text does not belong in section 10, as section 10
>>>> is about what needs to be documented to define an INFO 
>> package, not 
>>>> about what the implementor actually needs to do.
>>>> It actually relates to whether it was appropriate to 
>> select INFO in 
>>>> the first place to do the job, as if the information transfer 
>>>> requirements are inappropriate to the underlying transport 
>>>> capabilities of SIP, then INFO should not have been 
>> selected in the 
>>>> first place.
>>>>>    
>>>> On 5/5/10 8:28 AM, Christer Holmberg responded:
>>>>> So, I guess it should be somewhere in section 7.4.
>>>>>    
>>>> On 5/5/10 8:45 AM, DRAGE, Keith (Keith) replied:
>>>>> Yes, but obviously not in the same words. It needs to be
>>>> tailored as do not choose an INFO solution if you need to do this.
>>>>>    
>>>> I would argue that section 7.3 already goes over this 
>>> ground in terms 
>>>> of whether INFO is appropriate for a particular use case. And, in 
>>>> fact,
>>>> *most* of the proposed text isn't talking about what 
>> event package 
>>>> designers should do. Some of it does, but for the most 
>>> part, the text 
>>>> talks about what implementors are required to do. The 
>>> document *does* 
>>>> have a section that covers this ground: section 3.
>>>>
>>>> To that end, I would think we could split this into two 
>>> sections. The 
>>>> part dealing with implementation issues would be a new 
>> section 3.5; 
>>>> the other would form an introduction to the existing section 10.9.
>>>>
>>>> Incorporating the other suggestions made so far, then, I 
>>> would propose 
>>>> the following sections for the INFO document:
>>>>
>>>>> 3.5.  Rate and Size of INFO Requests
>>>>>
>>>>>      INFO messages differ from many other sorts of SIP 
>> messages in
>>>>>      that they carry application information, and the 
>>> size and rate
>>>>>      of the INFO message is directly determined by the 
>>> application.
>>>>>      This can cause application information traffic to interfere
>>>>>      with other traffic on that infrastructure, or to 
>>> self-interfere
>>>>>      when data rates become too high.
>>>>>
>>>>>      Implementors need to consider the impact that the 
>>> size and the
>>>>>      rate of the INFO messages have on the network and on other
>>>>>      traffic, since it normally cannot be ensured that 
>>> INFO messages
>>>>>      will be carried over a congestion-controlled 
>>> transport protocol
>>>>>      end-to-end.  Even if an INFO message is sent over such a
>>>>>      transport protocol, a downstream SIP entity might 
>> forward the
>>>>>      message over a transport protocol that does not provide
>>>>>      congestion control.
>>>>>
>>>>>      Therefore, an application SHOULD generally have at most one
>>>>>      outstanding INFO message.  Implementors are reminded 
>>> to observe
>>>>>      the guidelines in RFC3261 Section 18 regarding the 
>>> size of SIP
>>>>>      messages, such as the guideline to not exceed message sizes
>>>>>      of 1300 bytes when the path MTU is unknown. 
>> RFC5405 provides
>>>>>      additional guidelines for applications using the 
>>> User Datagram
>>>>>      Protocol (UDP) that may be useful background reading.
>>>>>
>>>>>
>>>>>
>>>>> 10.9 Rate of INFO Requests
>>>>>
>>>>>      A designer of an Info package needs to be mindful of the
>>>>>      potential congestion impact their application can 
>>> have on both
>>>>>      the SIP network and the Internet at large.  See section 3.5
>>>>>      for a discussion of congestion-related issues.
>>>>>
>>>>>      If there is a maximum or minimum rate at which UAs can send
>>>>>      INFO requests associated with the Info Package 
>>> within a dialog,
>>>>>      the Info Package specification MUST document the 
>> rate values.
>>>>>      If the rates can vary, the Info Package specification MAY
>>>>>      define Info Package parameters that UAs can use to indicate
>>>>>      or negotiate the rates.  Alternatively the rate information
>>>>>      can be part of the application data information associated
>>>>>      with the Info Package.
>>>>
>>>> Does that sound reasonable?
>>>>
>>>> /a
>>>>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 

From keith.drage@alcatel-lucent.com  Mon May 17 13:00:38 2010
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
Date: Mon, 17 May 2010 21:59:30 +0200
Thread-Topic: [sipcore] Call for Consensus: INFO Specification and Congestion
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Cc: Adam Roach <adam@estacado.net>, SIPCORE <sipcore@ietf.org>, "lars.eggert@nokia.com" <lars.eggert@nokia.com>
Subject: Re: [sipcore] Call for Consensus: INFO Specification and Congestion
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If the transport protocol is used by the application protocol in a manner t=
hat can get messages out of sequence, then the application protocol needs t=
o provide its own capabilities to resolve the issue.

So yes, when using INFO as a compelled signalling mechanism, it is reliable=
.

If you do not use it that way, then the application has to put in its own s=
equence numbers, or whatever mechanism it so desires, to achieve the same p=
urpose. That is of course assuming that the application protocol vulnerable=
 to dropped or out of sequence messages in the first place.=20

I believe I suggested some time ago on a very early draft that we might wan=
t to put something about this in the document, but no one ever took it up. =
Too late now.

regards

Keith

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20
> Sent: Monday, May 17, 2010 3:54 PM
> To: DRAGE, Keith (Keith)
> Cc: Christer Holmberg; Adam Roach; Adam Roach; SIPCORE;=20
> lars.eggert@nokia.com
> Subject: Re: [sipcore] Call for Consensus: INFO Specification=20
> and Congestion
>=20
> Maybe this is moot because Keith has agreed to the latest proposal...
>=20
> I was wondering about your objection to recommending at most=20
> one outstanding INFO message - because there are useful=20
> things to do with this.
>=20
> In general, having multiple outstanding requests in a dialog=20
> is problematic. With retransmission of lost messages this can=20
> result in some messages being rejected due to CSeq errors.=20
> While this isn't necessarily fatal, and there can be a few=20
> cases where this can still have utility, those cases are rare.
>=20
> 	Thanks,
> 	Paul
>=20
> DRAGE, Keith (Keith) wrote:
> > I'm OK with this proposal.
> >=20
> > Keith
> >=20
> >> -----Original Message-----
> >> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> >> Sent: Monday, May 17, 2010 6:32 AM
> >> To: DRAGE, Keith (Keith); Adam Roach
> >> Cc: Adam Roach; SIPCORE; lars.eggert@nokia.com
> >> Subject: RE: [sipcore] Call for Consensus: INFO Specification and=20
> >> Congestion
> >>
> >>
> >> Hi,
> >>
> >> I appologize that it has taken me a while to reply on this issue.=20
> >> I've been busy with 3GPP the last few weeks.
> >>
> >>
> >> Note that the draft already contains section 7.3, which=20
> talks about=20
> >> rate and volume. Nobody has commented on that, so I assume=20
> it is ok.
> >>
> >> My suggestion would be to:
> >>
> >> -	Remove section 7.3
> >> -	Add new section 7.4.1.2, that talks about rate and volume
> >> -	New section 7.4.1.2 consists of text from old section=20
> >> 7.3 and the suggested new text for section 10.9.
> >> -	Keep section 10.9 unchanged (draft version -07)
> >>
> >> So, new section 7.4.1.2 could look like:
> >>
> >>
> >> "7.4.1.2.  INFO Request Rate and Volume
> >>
> >>    INFO messages differ from many other sorts of SIP=20
> messages in that
> >>    they carry application information, and the size and rate of the
> >>    INFO message is directly determined by the application. This can
> >>    cause application information traffic to interfere with other=20
> >> traffic
> >>    on that infrastructure, or to self-interfere when data=20
> rates become
> >>    too high.
> >>
> >>    There is no default throttling mechanism for INFO=20
> requests. Apart
> >>    from the SIP session establishment, the number of SIP messages
> >>    exchanged during the lifetime a normal SIP session is=20
> rather small.
> >>
> >>    Some applications, like sending of DTMF tones, can=20
> generate a burst
> >>    of up to 20 messages per second.  Other applications,=20
> like constant
> >>    GPS location updates, could generate a high rate of=20
> INFO requests
> >>    during the lifetime of the invite dialog usage.
> >>
> >>    A designer of an INFO package, and the application that uses it,
> >>    need to consider the impact that the size and the rate=20
> of the INFO
> >>    messages have on the network and on other traffic, since it=20
> >> normally
> >>    cannot be ensured that INFO messages will be carried over a
> >>    congestion-controlled transport protocol end-to-end. Even if an
> >>    INFO message is sent over a such a transport protocol,=20
> a downstream
> >>    SIP entity might forward the message over a transport=20
> protocol that
> >>    does not provide congestion control.
> >>
> >>    Furthermore, SIP messages tend to be relatively small,=20
> on the order
> >>    of 500 Bytes to 32K Bytes. SIP is a poor mechanism for direct
> >>    exchange of bulk data beyond these limits, especially if the=20
> >> headers
> >>    plus body exceed the UDP MTU [RFC0768]. Appropriate=20
> mechanisms for
> >>    such traffic include HTTP [RFC2616], MSRP [RFC4975], or=20
> other media
> >>    plane data transport mechanisms.
> >>
> >>    RFC5405 provides additional guidelines for applications=20
> using the=20
> >>    User Datagram Protocol (UDP) that may be useful background=20
> >> reading."
> >>
> >> And, section 10.9 would look like:
> >>
> >> "10.9.  Rate of INFO Requests
> >>
> >>    If there is a maximum or minimum rate at which UAs can send INFO
> >>    requests associated with the Info Package within a=20
> dialog, the Info
> >>    Package specification MUST document the rate values.
> >>
> >>    If the rates can vary, the Info Package specification=20
> MAY define=20
> >> Info
> >>    Package parameters that UAs can use to indicate or negotiate the
> >>    rates.  Alternatively the rate information can be part of the
> >>    application data information associated with the Info Package."
> >>
> >>
> >> Would people be ok with that?
> >>
> >> Regards,
> >>
> >> Christer
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >> =20
> >>
> >>> -----Original Message-----
> >>> From: DRAGE, Keith (Keith) [mailto:keith.drage@alcatel-lucent.com]
> >>> Sent: 17. toukokuuta 2010 1:03
> >>> To: Adam Roach
> >>> Cc: Christer Holmberg; Adam Roach; SIPCORE; lars.eggert@nokia.com
> >>> Subject: RE: [sipcore] Call for Consensus: INFO Specification and=20
> >>> Congestion
> >>>
> >>> This proposed paragraph still causes me issues:
> >>>
> >>>>      Therefore, an application SHOULD generally have at most one
> >>>>      outstanding INFO message.  Implementors are reminded
> >> to observe
> >>>>      the guidelines in RFC3261 Section 18 regarding the
> >> size of SIP
> >>>>      messages, such as the guideline to not exceed message sizes
> >>>>      of 1300 bytes when the path MTU is unknown. RFC5405 provides
> >>>>      additional guidelines for applications using the
> >> User Datagram
> >>>>      Protocol (UDP) that may be useful background reading.
> >>> I already objected to the first sentence above as I do not
> >> believe it
> >>> solves the issue, and removes useful capabilities that may
> >> be desired
> >>> by some applications. I CANNOT accept this sentence.
> >>>
> >>> The second sentence above is an inexact extraction, and as such=20
> >>> misleads completely. The correct quote is below:
> >>>
> >>>    If a request is within 200 bytes of the path MTU, or if it is=20
> >>> larger
> >>>    than 1300 bytes and the path MTU is unknown, the=20
> request MUST be=20
> >>> sent
> >>>    using an RFC 2914 [43] congestion controlled transport=20
> protocol,=20
> >>> such
> >>>    as TCP. If this causes a change in the transport
> >> protocol from the
> >>>    one indicated in the top Via, the value in the top Via MUST be
> >>>    changed.  This prevents fragmentation of messages over UDP and
> >>>    provides congestion control for larger messages.  However,
> >>>    implementations MUST be able to handle messages up to=20
> the maximum
> >>>    datagram packet size.  For UDP, this size is 65,535 bytes,=20
> >>> including
> >>>    IP and UDP headers.
> >>>
> >>> Nothing about not exceeding message sizes. Only restriction
> >> is to send
> >>> using a congestion controlled transport protocol if it=20
> exceeds MTU=20
> >>> size. Therefore second sentence as currently drafted is
> >> unacceptable.
> >>>
> >>> regards
> >>>
> >>> Keith
> >>> =20
> >>>
> >>>> -----Original Message-----
> >>>> From: Adam Roach [mailto:adam@nostrum.com]
> >>>> Sent: Thursday, May 13, 2010 8:12 PM
> >>>> To: DRAGE, Keith (Keith)
> >>>> Cc: Christer Holmberg; Adam Roach; SIPCORE; lars.eggert@nokia.com
> >>>> Subject: Re: [sipcore] Call for Consensus: INFO=20
> Specification and=20
> >>>> Congestion
> >>>>
> >>>> [as a participant]
> >>>>
> >>>> Regarding the proposed text that I sent to the mailing list
> >>> last week,
> >>>> there was an exchange regarding where in the document
> >> this behavior
> >>>> goes;
> >>>>
> >>>> On 5/2/10 6:57 AM, DRAGE, Keith (Keith) wrote:
> >>>>> ...[T]his text does not belong in section 10, as section 10
> >>>> is about what needs to be documented to define an INFO
> >> package, not
> >>>> about what the implementor actually needs to do.
> >>>> It actually relates to whether it was appropriate to
> >> select INFO in
> >>>> the first place to do the job, as if the information transfer=20
> >>>> requirements are inappropriate to the underlying transport=20
> >>>> capabilities of SIP, then INFO should not have been
> >> selected in the
> >>>> first place.
> >>>>>   =20
> >>>> On 5/5/10 8:28 AM, Christer Holmberg responded:
> >>>>> So, I guess it should be somewhere in section 7.4.
> >>>>>   =20
> >>>> On 5/5/10 8:45 AM, DRAGE, Keith (Keith) replied:
> >>>>> Yes, but obviously not in the same words. It needs to be
> >>>> tailored as do not choose an INFO solution if you need=20
> to do this.
> >>>>>   =20
> >>>> I would argue that section 7.3 already goes over this
> >>> ground in terms
> >>>> of whether INFO is appropriate for a particular use=20
> case. And, in=20
> >>>> fact,
> >>>> *most* of the proposed text isn't talking about what
> >> event package
> >>>> designers should do. Some of it does, but for the most
> >>> part, the text
> >>>> talks about what implementors are required to do. The
> >>> document *does*
> >>>> have a section that covers this ground: section 3.
> >>>>
> >>>> To that end, I would think we could split this into two
> >>> sections. The
> >>>> part dealing with implementation issues would be a new
> >> section 3.5;
> >>>> the other would form an introduction to the existing=20
> section 10.9.
> >>>>
> >>>> Incorporating the other suggestions made so far, then, I
> >>> would propose
> >>>> the following sections for the INFO document:
> >>>>
> >>>>> 3.5.  Rate and Size of INFO Requests
> >>>>>
> >>>>>      INFO messages differ from many other sorts of SIP
> >> messages in
> >>>>>      that they carry application information, and the
> >>> size and rate
> >>>>>      of the INFO message is directly determined by the
> >>> application.
> >>>>>      This can cause application information traffic to interfere
> >>>>>      with other traffic on that infrastructure, or to
> >>> self-interfere
> >>>>>      when data rates become too high.
> >>>>>
> >>>>>      Implementors need to consider the impact that the
> >>> size and the
> >>>>>      rate of the INFO messages have on the network and on other
> >>>>>      traffic, since it normally cannot be ensured that
> >>> INFO messages
> >>>>>      will be carried over a congestion-controlled
> >>> transport protocol
> >>>>>      end-to-end.  Even if an INFO message is sent over such a
> >>>>>      transport protocol, a downstream SIP entity might
> >> forward the
> >>>>>      message over a transport protocol that does not provide
> >>>>>      congestion control.
> >>>>>
> >>>>>      Therefore, an application SHOULD generally have at most one
> >>>>>      outstanding INFO message.  Implementors are reminded
> >>> to observe
> >>>>>      the guidelines in RFC3261 Section 18 regarding the
> >>> size of SIP
> >>>>>      messages, such as the guideline to not exceed message sizes
> >>>>>      of 1300 bytes when the path MTU is unknown.=20
> >> RFC5405 provides
> >>>>>      additional guidelines for applications using the
> >>> User Datagram
> >>>>>      Protocol (UDP) that may be useful background reading.
> >>>>>
> >>>>>
> >>>>>
> >>>>> 10.9 Rate of INFO Requests
> >>>>>
> >>>>>      A designer of an Info package needs to be mindful of the
> >>>>>      potential congestion impact their application can
> >>> have on both
> >>>>>      the SIP network and the Internet at large.  See section 3.5
> >>>>>      for a discussion of congestion-related issues.
> >>>>>
> >>>>>      If there is a maximum or minimum rate at which UAs can send
> >>>>>      INFO requests associated with the Info Package
> >>> within a dialog,
> >>>>>      the Info Package specification MUST document the
> >> rate values.
> >>>>>      If the rates can vary, the Info Package specification MAY
> >>>>>      define Info Package parameters that UAs can use to indicate
> >>>>>      or negotiate the rates.  Alternatively the rate information
> >>>>>      can be part of the application data information associated
> >>>>>      with the Info Package.
> >>>>
> >>>> Does that sound reasonable?
> >>>>
> >>>> /a
> >>>>
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
> >=20
> =

From christer.holmberg@ericsson.com  Mon May 17 21:25:16 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>, Paul Kyzivat <pkyzivat@cisco.com>
Date: Tue, 18 May 2010 06:22:53 +0200
Thread-Topic: [sipcore] Call for Consensus: INFO Specification and Congestion
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References: <4BDB52AB.9000507@estacado.net>, <EDC0A1AE77C57744B664A310A0B23AE212F4324D@FRMRSSXCHMBSC3.dc-m.alcatel-lucent.com> <FF84A09F50A6DC48ACB6714F4666CC74632A894305@ESESSCMS0354.eemea.ericsson.se> <EDC0A1AE77C57744B664A310A0B23AE212F4336F@FRMRSSXCHMBSC3.dc-m.alcatel-lucent.com> <FF84A09F50A6DC48ACB6714F4666CC74632B2DA6BC@ESESSCMS0354.eemea.ericsson.se> <EDC0A1AE77C57744B664A310A0B23AE2130E0216@FRMRSSXCHMBSC3.dc-m.alcatel-lucent.com> <4BEC4EEE.70008@nostrum.com> <EDC0A1AE77C57744B664A310A0B23AE2131CF251@FRMRSSXCHMBSC3.dc-m.alcatel-lucent.com> <FF84A09F50A6DC48ACB6714F4666CC7465C5262F8E@ESESSCMS0354.eemea.ericsson.se> <EDC0A1AE77C57744B664A310A0B23AE2131CF36B@FRMRSSXCHMBSC3.dc-m.alcatel-lucent.com> <4BF15888.1040905@cisco.com>, <EDC0A1AE77C57744B664A310A0B23AE2131CF5C1@FRMRSSXCHMBSC3.dc-m.alcatel-lucent.com>
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Cc: Adam Roach <adam@estacado.net>, SIPCORE <sipcore@ietf.org>, "lars.eggert@nokia.com" <lars.eggert@nokia.com>
Subject: Re: [sipcore] Call for Consensus: INFO Specification and Congestion
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Hi,

>If the transport protocol is used by the application protocol in a manner =
that can get messages out of sequence, then the application protocol needs =
to provide its own capabilities to resolve the issue.
>
>So yes, when using INFO as a compelled signalling mechanism, it is reliabl=
e.
>
>If you do not use it that way, then the application has to put in its own =
sequence numbers, or whatever mechanism it so desires, to achieve the same =
purpose. That is of course assuming that the application protocol vulnerabl=
e to dropped or out of sequence messages in the=20
>first place.
>
>I believe I suggested some time ago on a very early draft that we might wa=
nt to put something about this in the document, but no one ever took it up.=
 Too late now.

Some text does exist in Section 3.4.

Also, I don't believe we had the restriction in the old RFC, so in the even=
t that thera are existing implementations sending multiple outstanding INFO=
s, I don't want to suddenly make them non-standard compliant.

So, Lars, can you live with the text I suggested? :)

Regards,

Christer


> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Monday, May 17, 2010 3:54 PM
> To: DRAGE, Keith (Keith)
> Cc: Christer Holmberg; Adam Roach; Adam Roach; SIPCORE;
> lars.eggert@nokia.com
> Subject: Re: [sipcore] Call for Consensus: INFO Specification
> and Congestion
>
> Maybe this is moot because Keith has agreed to the latest proposal...
>
> I was wondering about your objection to recommending at most
> one outstanding INFO message - because there are useful
> things to do with this.
>
> In general, having multiple outstanding requests in a dialog
> is problematic. With retransmission of lost messages this can
> result in some messages being rejected due to CSeq errors.
> While this isn't necessarily fatal, and there can be a few
> cases where this can still have utility, those cases are rare.
>
>       Thanks,
>       Paul
>
> DRAGE, Keith (Keith) wrote:
> > I'm OK with this proposal.
> >
> > Keith
> >
> >> -----Original Message-----
> >> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> >> Sent: Monday, May 17, 2010 6:32 AM
> >> To: DRAGE, Keith (Keith); Adam Roach
> >> Cc: Adam Roach; SIPCORE; lars.eggert@nokia.com
> >> Subject: RE: [sipcore] Call for Consensus: INFO Specification and
> >> Congestion
> >>
> >>
> >> Hi,
> >>
> >> I appologize that it has taken me a while to reply on this issue.
> >> I've been busy with 3GPP the last few weeks.
> >>
> >>
> >> Note that the draft already contains section 7.3, which
> talks about
> >> rate and volume. Nobody has commented on that, so I assume
> it is ok.
> >>
> >> My suggestion would be to:
> >>
> >> -  Remove section 7.3
> >> -  Add new section 7.4.1.2, that talks about rate and volume
> >> -  New section 7.4.1.2 consists of text from old section
> >> 7.3 and the suggested new text for section 10.9.
> >> -  Keep section 10.9 unchanged (draft version -07)
> >>
> >> So, new section 7.4.1.2 could look like:
> >>
> >>
> >> "7.4.1.2.  INFO Request Rate and Volume
> >>
> >>    INFO messages differ from many other sorts of SIP
> messages in that
> >>    they carry application information, and the size and rate of the
> >>    INFO message is directly determined by the application. This can
> >>    cause application information traffic to interfere with other
> >> traffic
> >>    on that infrastructure, or to self-interfere when data
> rates become
> >>    too high.
> >>
> >>    There is no default throttling mechanism for INFO
> requests. Apart
> >>    from the SIP session establishment, the number of SIP messages
> >>    exchanged during the lifetime a normal SIP session is
> rather small.
> >>
> >>    Some applications, like sending of DTMF tones, can
> generate a burst
> >>    of up to 20 messages per second.  Other applications,
> like constant
> >>    GPS location updates, could generate a high rate of
> INFO requests
> >>    during the lifetime of the invite dialog usage.
> >>
> >>    A designer of an INFO package, and the application that uses it,
> >>    need to consider the impact that the size and the rate
> of the INFO
> >>    messages have on the network and on other traffic, since it
> >> normally
> >>    cannot be ensured that INFO messages will be carried over a
> >>    congestion-controlled transport protocol end-to-end. Even if an
> >>    INFO message is sent over a such a transport protocol,
> a downstream
> >>    SIP entity might forward the message over a transport
> protocol that
> >>    does not provide congestion control.
> >>
> >>    Furthermore, SIP messages tend to be relatively small,
> on the order
> >>    of 500 Bytes to 32K Bytes. SIP is a poor mechanism for direct
> >>    exchange of bulk data beyond these limits, especially if the
> >> headers
> >>    plus body exceed the UDP MTU [RFC0768]. Appropriate
> mechanisms for
> >>    such traffic include HTTP [RFC2616], MSRP [RFC4975], or
> other media
> >>    plane data transport mechanisms.
> >>
> >>    RFC5405 provides additional guidelines for applications
> using the
> >>    User Datagram Protocol (UDP) that may be useful background
> >> reading."
> >>
> >> And, section 10.9 would look like:
> >>
> >> "10.9.  Rate of INFO Requests
> >>
> >>    If there is a maximum or minimum rate at which UAs can send INFO
> >>    requests associated with the Info Package within a
> dialog, the Info
> >>    Package specification MUST document the rate values.
> >>
> >>    If the rates can vary, the Info Package specification
> MAY define
> >> Info
> >>    Package parameters that UAs can use to indicate or negotiate the
> >>    rates.  Alternatively the rate information can be part of the
> >>    application data information associated with the Info Package."
> >>
> >>
> >> Would people be ok with that?
> >>
> >> Regards,
> >>
> >> Christer
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>> -----Original Message-----
> >>> From: DRAGE, Keith (Keith) [mailto:keith.drage@alcatel-lucent.com]
> >>> Sent: 17. toukokuuta 2010 1:03
> >>> To: Adam Roach
> >>> Cc: Christer Holmberg; Adam Roach; SIPCORE; lars.eggert@nokia.com
> >>> Subject: RE: [sipcore] Call for Consensus: INFO Specification and
> >>> Congestion
> >>>
> >>> This proposed paragraph still causes me issues:
> >>>
> >>>>      Therefore, an application SHOULD generally have at most one
> >>>>      outstanding INFO message.  Implementors are reminded
> >> to observe
> >>>>      the guidelines in RFC3261 Section 18 regarding the
> >> size of SIP
> >>>>      messages, such as the guideline to not exceed message sizes
> >>>>      of 1300 bytes when the path MTU is unknown. RFC5405 provides
> >>>>      additional guidelines for applications using the
> >> User Datagram
> >>>>      Protocol (UDP) that may be useful background reading.
> >>> I already objected to the first sentence above as I do not
> >> believe it
> >>> solves the issue, and removes useful capabilities that may
> >> be desired
> >>> by some applications. I CANNOT accept this sentence.
> >>>
> >>> The second sentence above is an inexact extraction, and as such
> >>> misleads completely. The correct quote is below:
> >>>
> >>>    If a request is within 200 bytes of the path MTU, or if it is
> >>> larger
> >>>    than 1300 bytes and the path MTU is unknown, the
> request MUST be
> >>> sent
> >>>    using an RFC 2914 [43] congestion controlled transport
> protocol,
> >>> such
> >>>    as TCP. If this causes a change in the transport
> >> protocol from the
> >>>    one indicated in the top Via, the value in the top Via MUST be
> >>>    changed.  This prevents fragmentation of messages over UDP and
> >>>    provides congestion control for larger messages.  However,
> >>>    implementations MUST be able to handle messages up to
> the maximum
> >>>    datagram packet size.  For UDP, this size is 65,535 bytes,
> >>> including
> >>>    IP and UDP headers.
> >>>
> >>> Nothing about not exceeding message sizes. Only restriction
> >> is to send
> >>> using a congestion controlled transport protocol if it
> exceeds MTU
> >>> size. Therefore second sentence as currently drafted is
> >> unacceptable.
> >>>
> >>> regards
> >>>
> >>> Keith
> >>>
> >>>
> >>>> -----Original Message-----
> >>>> From: Adam Roach [mailto:adam@nostrum.com]
> >>>> Sent: Thursday, May 13, 2010 8:12 PM
> >>>> To: DRAGE, Keith (Keith)
> >>>> Cc: Christer Holmberg; Adam Roach; SIPCORE; lars.eggert@nokia.com
> >>>> Subject: Re: [sipcore] Call for Consensus: INFO
> Specification and
> >>>> Congestion
> >>>>
> >>>> [as a participant]
> >>>>
> >>>> Regarding the proposed text that I sent to the mailing list
> >>> last week,
> >>>> there was an exchange regarding where in the document
> >> this behavior
> >>>> goes;
> >>>>
> >>>> On 5/2/10 6:57 AM, DRAGE, Keith (Keith) wrote:
> >>>>> ...[T]his text does not belong in section 10, as section 10
> >>>> is about what needs to be documented to define an INFO
> >> package, not
> >>>> about what the implementor actually needs to do.
> >>>> It actually relates to whether it was appropriate to
> >> select INFO in
> >>>> the first place to do the job, as if the information transfer
> >>>> requirements are inappropriate to the underlying transport
> >>>> capabilities of SIP, then INFO should not have been
> >> selected in the
> >>>> first place.
> >>>>>
> >>>> On 5/5/10 8:28 AM, Christer Holmberg responded:
> >>>>> So, I guess it should be somewhere in section 7.4.
> >>>>>
> >>>> On 5/5/10 8:45 AM, DRAGE, Keith (Keith) replied:
> >>>>> Yes, but obviously not in the same words. It needs to be
> >>>> tailored as do not choose an INFO solution if you need
> to do this.
> >>>>>
> >>>> I would argue that section 7.3 already goes over this
> >>> ground in terms
> >>>> of whether INFO is appropriate for a particular use
> case. And, in
> >>>> fact,
> >>>> *most* of the proposed text isn't talking about what
> >> event package
> >>>> designers should do. Some of it does, but for the most
> >>> part, the text
> >>>> talks about what implementors are required to do. The
> >>> document *does*
> >>>> have a section that covers this ground: section 3.
> >>>>
> >>>> To that end, I would think we could split this into two
> >>> sections. The
> >>>> part dealing with implementation issues would be a new
> >> section 3.5;
> >>>> the other would form an introduction to the existing
> section 10.9.
> >>>>
> >>>> Incorporating the other suggestions made so far, then, I
> >>> would propose
> >>>> the following sections for the INFO document:
> >>>>
> >>>>> 3.5.  Rate and Size of INFO Requests
> >>>>>
> >>>>>      INFO messages differ from many other sorts of SIP
> >> messages in
> >>>>>      that they carry application information, and the
> >>> size and rate
> >>>>>      of the INFO message is directly determined by the
> >>> application.
> >>>>>      This can cause application information traffic to interfere
> >>>>>      with other traffic on that infrastructure, or to
> >>> self-interfere
> >>>>>      when data rates become too high.
> >>>>>
> >>>>>      Implementors need to consider the impact that the
> >>> size and the
> >>>>>      rate of the INFO messages have on the network and on other
> >>>>>      traffic, since it normally cannot be ensured that
> >>> INFO messages
> >>>>>      will be carried over a congestion-controlled
> >>> transport protocol
> >>>>>      end-to-end.  Even if an INFO message is sent over such a
> >>>>>      transport protocol, a downstream SIP entity might
> >> forward the
> >>>>>      message over a transport protocol that does not provide
> >>>>>      congestion control.
> >>>>>
> >>>>>      Therefore, an application SHOULD generally have at most one
> >>>>>      outstanding INFO message.  Implementors are reminded
> >>> to observe
> >>>>>      the guidelines in RFC3261 Section 18 regarding the
> >>> size of SIP
> >>>>>      messages, such as the guideline to not exceed message sizes
> >>>>>      of 1300 bytes when the path MTU is unknown.
> >> RFC5405 provides
> >>>>>      additional guidelines for applications using the
> >>> User Datagram
> >>>>>      Protocol (UDP) that may be useful background reading.
> >>>>>
> >>>>>
> >>>>>
> >>>>> 10.9 Rate of INFO Requests
> >>>>>
> >>>>>      A designer of an Info package needs to be mindful of the
> >>>>>      potential congestion impact their application can
> >>> have on both
> >>>>>      the SIP network and the Internet at large.  See section 3.5
> >>>>>      for a discussion of congestion-related issues.
> >>>>>
> >>>>>      If there is a maximum or minimum rate at which UAs can send
> >>>>>      INFO requests associated with the Info Package
> >>> within a dialog,
> >>>>>      the Info Package specification MUST document the
> >> rate values.
> >>>>>      If the rates can vary, the Info Package specification MAY
> >>>>>      define Info Package parameters that UAs can use to indicate
> >>>>>      or negotiate the rates.  Alternatively the rate information
> >>>>>      can be part of the application data information associated
> >>>>>      with the Info Package.
> >>>>
> >>>> Does that sound reasonable?
> >>>>
> >>>> /a
> >>>>
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
> >
>=

From gao.yang2@zte.com.cn  Tue May 18 01:27:34 2010
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Subject: [sipcore] Question about P-Associated-URI header's syntax defined in RFC3455
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Content-Type: text/plain; charset="US-ASCII"

Hi,

If people has/have discussed this issue in sip relating WG, sorry for it.

Yongping(Yongping Huang) and Jingning(Jingning He) find a inconsistent 
problem about P-Associated-URI header's syntax defined in RFC3455. By text 
description in 4.1.2.1 and 4.1.2.2(the highlight part), P-Associated-URI 
may have zero URI. But refer to the syntax description part in 5.1, it 
seams the header would have at least one URI. 

Is this an inconsistent problem?

Thanks,

Gao


//Jingning's correction proposal as:

P-Associated-URI       = "P-Associated-URI" HCOLON
                               [p-aso-uri-spec *(COMMA p-aso-uri-spec)]
or,
 
P-Associated-URI       = "P-Associated-URI" HCOLON
                               [p-aso-uri-spec ]*(COMMA p-aso-uri-spec) 
 

//Parts of text from RFC3455:

4.1.2.1 Procedures at the UA

   The header value contains a list of zero or more associated URIs to
   the address-of-record URI.  The UAC MAY use any of the associated
   URIs to populate the From header value, or any other SIP header value
   that provides information of the identity of the calling party, in a
   subsequent request.


4.1.2.2 Procedures at the registrar

   In case the address-of-record under registration does not have any
   other SIP or SIPS URIs associated, the registrar MUST include an
   empty P-Associated-URI header value.


5.1 P-Associated-URI header syntax

   The syntax of the P-Associated-URI header is described as follows:

      P-Associated-URI       = "P-Associated-URI" HCOLON
                               (p-aso-uri-spec)
                               *(COMMA p-aso-uri-spec)





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<br><font size=2 face="sans-serif">Hi,</font>
<br>
<br><font size=2 face="sans-serif">If people has/have discussed this issue
in sip relating WG, sorry for it.</font>
<br>
<br><font size=2 face="sans-serif">Yongping(Yongping Huang) and Jingning(Jingning
He) find a inconsistent problem about P-Associated-URI header's syntax
defined in RFC3455. By text description in 4.1.2.1 and 4.1.2.2(the highlight
part), P-Associated-URI may have zero URI. But refer to the syntax description
part in 5.1, it seams the header would have at least one URI. </font>
<br>
<br><font size=2 face="sans-serif">Is this an inconsistent problem?</font>
<br>
<br><font size=2 face="sans-serif">Thanks,</font>
<br>
<br><font size=2 face="sans-serif">Gao</font>
<br>
<br>
<br><font size=2 face="sans-serif">//Jingning's correction proposal as:</font>
<br>
<br><font size=2 face="sans-serif">P-Associated-URI &nbsp; &nbsp; &nbsp;
= &quot;P-Associated-URI&quot; HCOLON</font>
<br><font size=2 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;[p-aso-uri-spec
*(COMMA p-aso-uri-spec)]</font>
<br><font size=2 face="sans-serif">or,</font>
<br><font size=2 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;</font>
<br><font size=2 face="sans-serif">P-Associated-URI &nbsp; &nbsp; &nbsp;
= &quot;P-Associated-URI&quot; HCOLON</font>
<br><font size=2 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;[p-aso-uri-spec
]*(COMMA p-aso-uri-spec) &nbsp; </font>
<br><font size=2 face="sans-serif">&nbsp; &nbsp;</font>
<br>
<br><font size=2 face="sans-serif">//Parts of text from RFC3455:</font>
<br>
<br><font size=2 face="sans-serif">4.1.2.1 Procedures at the UA</font>
<br>
<br><font size=2 face="sans-serif">&nbsp; &nbsp;The header value contains
a list of </font><font size=2 color=blue face="sans-serif">zero or more
associated URIs</font><font size=2 face="sans-serif"> to</font>
<br><font size=2 face="sans-serif">&nbsp; &nbsp;the address-of-record URI.
&nbsp;The UAC MAY use any of the associated</font>
<br><font size=2 face="sans-serif">&nbsp; &nbsp;URIs to populate the From
header value, or any other SIP header value</font>
<br><font size=2 face="sans-serif">&nbsp; &nbsp;that provides information
of the identity of the calling party, in a</font>
<br><font size=2 face="sans-serif">&nbsp; &nbsp;subsequent request.</font>
<br>
<br>
<br><font size=2 face="sans-serif">4.1.2.2 Procedures at the registrar</font>
<br>
<br><font size=2 face="sans-serif">&nbsp; &nbsp;In case the address-of-record
under registration does not have any</font>
<br><font size=2 face="sans-serif">&nbsp; &nbsp;other SIP or SIPS URIs
associated, the registrar MUST include </font><font size=2 color=blue face="sans-serif">an</font>
<br><font size=2 color=blue face="sans-serif">&nbsp; &nbsp;empty P-Associated-URI
header</font><font size=2 face="sans-serif"> value.</font>
<br>
<br>
<br><font size=2 face="sans-serif">5.1 P-Associated-URI header syntax</font>
<br>
<br><font size=2 face="sans-serif">&nbsp; &nbsp;The syntax of the P-Associated-URI
header is described as follows:</font>
<br>
<br><font size=2 face="sans-serif">&nbsp; &nbsp; &nbsp; P-Associated-URI
&nbsp; &nbsp; &nbsp; = &quot;P-Associated-URI&quot; HCOLON</font>
<br><font size=2 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;(p-aso-uri-spec)</font>
<br><font size=2 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;*(COMMA
p-aso-uri-spec)</font>
<br>
<br>
<br>
<br>
<br>
<br><font size=2 face="sans-serif">===================================<br>
 Zip &nbsp; &nbsp;: 210012<br>
 Tel &nbsp; &nbsp;: 87211<br>
 Tel2 &nbsp; :(+86)-025-52877211<br>
 e_mail : gao.yang2@zte.com.cn<br>
===================================</font><br><pre>
--------------------------------------------------------
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From adam@nostrum.com  Tue May 18 06:47:05 2010
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Cc: "DRAGE, Keith \(Keith\)" <keith.drage@alcatel-lucent.com>, Adam Roach <adam@estacado.net>, SIPCORE <sipcore@ietf.org>, "lars.eggert@nokia.com" <lars.eggert@nokia.com>
Subject: Re: [sipcore] Call for Consensus: INFO Specification and Congestion
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[as participant]

On 5/17/10 23:22, May 17, Christer Holmberg wrote:
> Also, I don't believe we had the restriction in the old RFC, so in the event that thera are existing implementations sending multiple outstanding INFOs, I don't want to suddenly make them non-standard compliant.
>    

They also don't include an Info-Package header field. No matter what you 
do, RFC 2976 implementations aren't going to be compliant to the new 
specification. Any arguments about "making RFC 2976 implementations 
non-compliant" are red herrings.

/a

From christer.holmberg@ericsson.com  Tue May 18 11:41:09 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Adam Roach <adam@nostrum.com>
Date: Tue, 18 May 2010 15:58:25 +0200
Thread-Topic: [sipcore] Call for Consensus: INFO Specification and Congestion
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Cc: "DRAGE, Keith \(Keith\)" <keith.drage@alcatel-lucent.com>, Adam Roach <adam@estacado.net>, SIPCORE <sipcore@ietf.org>, "lars.eggert@nokia.com" <lars.eggert@nokia.com>
Subject: Re: [sipcore] Call for Consensus: INFO Specification and Congestion
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Hi,

>>Also, I don't believe we had the restriction in the old RFC, so in the ev=
ent that thera are existing >>implementations sending multiple outstanding =
INFOs, I don't want to suddenly make them non-standard >>compliant.
>
>They also don't include an Info-Package header field. No matter what you
>do, RFC 2976 implementations aren't going to be compliant to the new
>specification. Any arguments about "making RFC 2976 implementations
>non-compliant" are red herrings.

The draft also defines the "legacy INFO usage", which means RFC 2976 implem=
entations will continue to be compliant.

Anyway, Lars has indicated that he can live with my suggested text, so my i=
ntention is to update the draft based on that.

Regards,

Christer=

From gao.yang2@zte.com.cn  Tue May 18 19:36:07 2010
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Hi,

In the address line of your mail, there's the mail list's address. But 
your mail is not in the archive. I suggest you chech your internal mail 
server and request Oscar to check the mail list's server for your e-mail 
address.

Contents inlines.

Thanks,

Gao 

> In 4.5.  Race Conditions and Target Refreshes, 
>    Additionally, UAs MUST treat
>    the exchange of a 2xx response to an INVITE request and its
>    corresponding ACK request as an offer/answer exchange.
> 
> What do above descriptions require to UAS?

This requires UAC and UAS (UAs). And the specific demand is to treate such 
messages as if with O/A.

> 
> >In my understanding, it is crossing case, while not glare one.
> 
> The figure in the previous mail is a glare case of UPDATE.
> And the crossing case of UPDATE is shown as below.
> 
>     A                               B
>     |                               |
>     |re-INV (offer1)                |
>     |------------------------------>|
>     |                   18x(answer1)|
>     |<------------------------------|
>     |                        non-2xx|
>     |<------------------------------|
>     |                               |
>     |                   /---------->|*
>     |ACK               /            |
>     |-----------------/------------>|
>     |                /              |
>     |UPDATE (offer2)/               |
>     |--------------/                |
>     |                               |
> 
> UAS will not be able to recognize the difference between
> the glare case and the crossing case.
> 

Yes, UAS can not recognize the difference. So, this use case has 
ambiguity, and in UAS's view, this is a glare case.
And the UAS would send 491, the session state would be the same in UAC and 
UAS's view.

I'd like to put your glare case further:

UAC send UPDATE in scope of Re-INVITE in it's view(before non-2xx of this 
Re-INVITE). But UAS get the UPDATE after the non-2xx(Re-INVITE)'s ACK.
UAC's view: UPDATE is in scope of the Re-INVITE;
UAS's view: UPDATE is out scope of the Re-INVITE.
For this case, the text suggest UAC send a new Re-INVITE/UPDATE after the 
2xx of UPDATE. This is defined in 3.5:

If
   the UAC of a re-INVITE transaction sends an UPDATE request with an
   offer, receives a non-2xx response to the re-INVITE, and then a 2xx
   response to the UPDATE request, the UA SHOULD generate a new re-
   INVITE or UPDATE request in order to make sure that both UAs have a
   common view of the state of the session, as described in Section 3.4.
 

--------------------------------------------------------
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Content-Type: text/html; charset="US-ASCII"


<br><font size=2 face="sans-serif">Hi,</font>
<br>
<br><font size=2 face="sans-serif">In the address line of your mail, there's
the mail list's address. But your mail is not in the archive. I suggest
you chech your internal mail server and request Oscar to check the mail
list's server for your e-mail address.</font>
<br>
<br><font size=2 face="sans-serif">Contents inlines.</font>
<br>
<br><font size=2 face="sans-serif">Thanks,</font>
<br>
<br><font size=2 face="sans-serif">Gao &nbsp;</font>
<br><tt><font size=2><br>
&gt; In 4.5. &nbsp;Race Conditions and Target Refreshes, <br>
&gt; &nbsp; &nbsp;Additionally, UAs MUST treat<br>
&gt; &nbsp; &nbsp;the exchange of a 2xx response to an INVITE request and
its<br>
&gt; &nbsp; &nbsp;corresponding ACK request as an offer/answer exchange.<br>
&gt; <br>
&gt; What do above descriptions require to UAS?</font></tt>
<br>
<br><tt><font size=2>This requires UAC and UAS (UAs). And the specific
demand is to treate such messages as if with O/A.</font></tt>
<br><tt><font size=2><br>
&gt; <br>
&gt; &gt;In my understanding, it is crossing case, while not glare one.<br>
&gt; <br>
&gt; The figure in the previous mail is a glare case of UPDATE.<br>
&gt; And the crossing case of UPDATE is shown as below.<br>
&gt; <br>
&gt; &nbsp; &nbsp; A &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; B<br>
&gt; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |<br>
&gt; &nbsp; &nbsp; |re-INV (offer1) &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp;|<br>
&gt; &nbsp; &nbsp; |------------------------------&gt;|<br>
&gt; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; 18x(answer1)|<br>
&gt; &nbsp; &nbsp; |&lt;------------------------------|<br>
&gt; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp;non-2xx|<br>
&gt; &nbsp; &nbsp; |&lt;------------------------------|<br>
&gt; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |<br>
&gt; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; /----------&gt;|*<br>
&gt; &nbsp; &nbsp; |ACK &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
/ &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|<br>
&gt; &nbsp; &nbsp; |-----------------/------------&gt;|<br>
&gt; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;/
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|<br>
&gt; &nbsp; &nbsp; |UPDATE (offer2)/ &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; |<br>
&gt; &nbsp; &nbsp; |--------------/ &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp;|<br>
&gt; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |<br>
&gt; <br>
&gt; UAS will not be able to recognize the difference between<br>
&gt; the glare case and the crossing case.<br>
&gt; </font></tt>
<br>
<br><tt><font size=2>Yes, UAS can not recognize the difference. So, this
use case has ambiguity, and in UAS's view, this is a glare case.</font></tt>
<br><tt><font size=2>And the UAS would send 491, the session state would
be the same in UAC and UAS's view.</font></tt>
<br>
<br><tt><font size=2>I'd like to put your glare case further:</font></tt>
<br>
<br><tt><font size=2>UAC send UPDATE in scope of Re-INVITE in it's view(before
non-2xx of this Re-INVITE). But UAS get the UPDATE after the non-2xx(Re-INVITE)'s
ACK.</font></tt>
<br><tt><font size=2>UAC's view: UPDATE is in scope of the Re-INVITE;</font></tt>
<br><tt><font size=2>UAS's view: UPDATE is out scope of the Re-INVITE.</font></tt>
<br><tt><font size=2>For this case, the text suggest UAC send a new Re-INVITE/UPDATE
after the 2xx of UPDATE. This is defined in 3.5:</font></tt>
<br>
<br><font size=2 face="Courier New">If</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp;the UAC of a re-INVITE
transaction sends an UPDATE request with an</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp;offer, receives a non-2xx
response to the re-INVITE, and then a 2xx</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp;response to the UPDATE
request, the UA SHOULD generate a new re-</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp;INVITE or UPDATE request
in order to make sure that both UAs have a</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp;common view of the state
of the session, as described in Section 3.4.</font>
<br><tt><font size=2>&nbsp;</font></tt><br><pre>
--------------------------------------------------------
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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           : Session Initiation Protocol (SIP) INFO Method and Package Framework
	Author(s)       : C. Holmberg, et al.
	Filename        : draft-ietf-sipcore-info-events-08.txt
	Pages           : 38
	Date            : 2010-05-19

This document defines a method, INFO, for the Session Initiation
Protocol (SIP), and an Info Package mechanism.  The document
obsoletes RFC 2976.  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 RFC 2976, referred to
as "legacy INFO Usage" in this document.

A URL for this Internet-Draft is:
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Cc: Adam Roach <adam@estacado.net>, SIPCORE <sipcore@ietf.org>, "lars.eggert@nokia.com" <lars.eggert@nokia.com>
Subject: Re: [sipcore] Call for Consensus: INFO Specification and Congestion
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DRAGE, Keith (Keith) wrote:
> If the transport protocol is used by the application protocol in a manner that can get messages out of sequence, then the application protocol needs to provide its own capabilities to resolve the issue.

In general there is little/nothing the application can do to ensure that 
  multiple in-flight INFO messages stay in order. Even if TCP is used on 
every hop, messages may be reordered as they transit a proxy. Hence 500 
responses due to the out of order messages may be received.

And yes, the application protocol can be designed to function in the 
face of that. I'm not suggesting that having multiple INFOs outstanding 
be forbidden - just reminding that there are more problems with using 
them than simply congestion.

> So yes, when using INFO as a compelled signalling mechanism, it is reliable.

I don't understand the above. What is a "compelled signalling mechanism"?

> If you do not use it that way, then the application has to put in its own sequence numbers, or whatever mechanism it so desires, to achieve the same purpose. That is of course assuming that the application protocol vulnerable to dropped or out of sequence messages in the first place. 
> 
> I believe I suggested some time ago on a very early draft that we might want to put something about this in the document, but no one ever took it up. Too late now.

As long as we are all on the same page about this I'm satisfied.

	Thanks,
	Paul

> regards
> 
> Keith
> 
>> -----Original Message-----
>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
>> Sent: Monday, May 17, 2010 3:54 PM
>> To: DRAGE, Keith (Keith)
>> Cc: Christer Holmberg; Adam Roach; Adam Roach; SIPCORE; 
>> lars.eggert@nokia.com
>> Subject: Re: [sipcore] Call for Consensus: INFO Specification 
>> and Congestion
>>
>> Maybe this is moot because Keith has agreed to the latest proposal...
>>
>> I was wondering about your objection to recommending at most 
>> one outstanding INFO message - because there are useful 
>> things to do with this.
>>
>> In general, having multiple outstanding requests in a dialog 
>> is problematic. With retransmission of lost messages this can 
>> result in some messages being rejected due to CSeq errors. 
>> While this isn't necessarily fatal, and there can be a few 
>> cases where this can still have utility, those cases are rare.
>>
>> 	Thanks,
>> 	Paul
>>
>> DRAGE, Keith (Keith) wrote:
>>> I'm OK with this proposal.
>>>
>>> Keith
>>>
>>>> -----Original Message-----
>>>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
>>>> Sent: Monday, May 17, 2010 6:32 AM
>>>> To: DRAGE, Keith (Keith); Adam Roach
>>>> Cc: Adam Roach; SIPCORE; lars.eggert@nokia.com
>>>> Subject: RE: [sipcore] Call for Consensus: INFO Specification and 
>>>> Congestion
>>>>
>>>>
>>>> Hi,
>>>>
>>>> I appologize that it has taken me a while to reply on this issue. 
>>>> I've been busy with 3GPP the last few weeks.
>>>>
>>>>
>>>> Note that the draft already contains section 7.3, which 
>> talks about 
>>>> rate and volume. Nobody has commented on that, so I assume 
>> it is ok.
>>>> My suggestion would be to:
>>>>
>>>> -	Remove section 7.3
>>>> -	Add new section 7.4.1.2, that talks about rate and volume
>>>> -	New section 7.4.1.2 consists of text from old section 
>>>> 7.3 and the suggested new text for section 10.9.
>>>> -	Keep section 10.9 unchanged (draft version -07)
>>>>
>>>> So, new section 7.4.1.2 could look like:
>>>>
>>>>
>>>> "7.4.1.2.  INFO Request Rate and Volume
>>>>
>>>>    INFO messages differ from many other sorts of SIP 
>> messages in that
>>>>    they carry application information, and the size and rate of the
>>>>    INFO message is directly determined by the application. This can
>>>>    cause application information traffic to interfere with other 
>>>> traffic
>>>>    on that infrastructure, or to self-interfere when data 
>> rates become
>>>>    too high.
>>>>
>>>>    There is no default throttling mechanism for INFO 
>> requests. Apart
>>>>    from the SIP session establishment, the number of SIP messages
>>>>    exchanged during the lifetime a normal SIP session is 
>> rather small.
>>>>    Some applications, like sending of DTMF tones, can 
>> generate a burst
>>>>    of up to 20 messages per second.  Other applications, 
>> like constant
>>>>    GPS location updates, could generate a high rate of 
>> INFO requests
>>>>    during the lifetime of the invite dialog usage.
>>>>
>>>>    A designer of an INFO package, and the application that uses it,
>>>>    need to consider the impact that the size and the rate 
>> of the INFO
>>>>    messages have on the network and on other traffic, since it 
>>>> normally
>>>>    cannot be ensured that INFO messages will be carried over a
>>>>    congestion-controlled transport protocol end-to-end. Even if an
>>>>    INFO message is sent over a such a transport protocol, 
>> a downstream
>>>>    SIP entity might forward the message over a transport 
>> protocol that
>>>>    does not provide congestion control.
>>>>
>>>>    Furthermore, SIP messages tend to be relatively small, 
>> on the order
>>>>    of 500 Bytes to 32K Bytes. SIP is a poor mechanism for direct
>>>>    exchange of bulk data beyond these limits, especially if the 
>>>> headers
>>>>    plus body exceed the UDP MTU [RFC0768]. Appropriate 
>> mechanisms for
>>>>    such traffic include HTTP [RFC2616], MSRP [RFC4975], or 
>> other media
>>>>    plane data transport mechanisms.
>>>>
>>>>    RFC5405 provides additional guidelines for applications 
>> using the 
>>>>    User Datagram Protocol (UDP) that may be useful background 
>>>> reading."
>>>>
>>>> And, section 10.9 would look like:
>>>>
>>>> "10.9.  Rate of INFO Requests
>>>>
>>>>    If there is a maximum or minimum rate at which UAs can send INFO
>>>>    requests associated with the Info Package within a 
>> dialog, the Info
>>>>    Package specification MUST document the rate values.
>>>>
>>>>    If the rates can vary, the Info Package specification 
>> MAY define 
>>>> Info
>>>>    Package parameters that UAs can use to indicate or negotiate the
>>>>    rates.  Alternatively the rate information can be part of the
>>>>    application data information associated with the Info Package."
>>>>
>>>>
>>>> Would people be ok with that?
>>>>
>>>> Regards,
>>>>
>>>> Christer
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>  
>>>>
>>>>> -----Original Message-----
>>>>> From: DRAGE, Keith (Keith) [mailto:keith.drage@alcatel-lucent.com]
>>>>> Sent: 17. toukokuuta 2010 1:03
>>>>> To: Adam Roach
>>>>> Cc: Christer Holmberg; Adam Roach; SIPCORE; lars.eggert@nokia.com
>>>>> Subject: RE: [sipcore] Call for Consensus: INFO Specification and 
>>>>> Congestion
>>>>>
>>>>> This proposed paragraph still causes me issues:
>>>>>
>>>>>>      Therefore, an application SHOULD generally have at most one
>>>>>>      outstanding INFO message.  Implementors are reminded
>>>> to observe
>>>>>>      the guidelines in RFC3261 Section 18 regarding the
>>>> size of SIP
>>>>>>      messages, such as the guideline to not exceed message sizes
>>>>>>      of 1300 bytes when the path MTU is unknown. RFC5405 provides
>>>>>>      additional guidelines for applications using the
>>>> User Datagram
>>>>>>      Protocol (UDP) that may be useful background reading.
>>>>> I already objected to the first sentence above as I do not
>>>> believe it
>>>>> solves the issue, and removes useful capabilities that may
>>>> be desired
>>>>> by some applications. I CANNOT accept this sentence.
>>>>>
>>>>> The second sentence above is an inexact extraction, and as such 
>>>>> misleads completely. The correct quote is below:
>>>>>
>>>>>    If a request is within 200 bytes of the path MTU, or if it is 
>>>>> larger
>>>>>    than 1300 bytes and the path MTU is unknown, the 
>> request MUST be 
>>>>> sent
>>>>>    using an RFC 2914 [43] congestion controlled transport 
>> protocol, 
>>>>> such
>>>>>    as TCP. If this causes a change in the transport
>>>> protocol from the
>>>>>    one indicated in the top Via, the value in the top Via MUST be
>>>>>    changed.  This prevents fragmentation of messages over UDP and
>>>>>    provides congestion control for larger messages.  However,
>>>>>    implementations MUST be able to handle messages up to 
>> the maximum
>>>>>    datagram packet size.  For UDP, this size is 65,535 bytes, 
>>>>> including
>>>>>    IP and UDP headers.
>>>>>
>>>>> Nothing about not exceeding message sizes. Only restriction
>>>> is to send
>>>>> using a congestion controlled transport protocol if it 
>> exceeds MTU 
>>>>> size. Therefore second sentence as currently drafted is
>>>> unacceptable.
>>>>> regards
>>>>>
>>>>> Keith
>>>>>  
>>>>>
>>>>>> -----Original Message-----
>>>>>> From: Adam Roach [mailto:adam@nostrum.com]
>>>>>> Sent: Thursday, May 13, 2010 8:12 PM
>>>>>> To: DRAGE, Keith (Keith)
>>>>>> Cc: Christer Holmberg; Adam Roach; SIPCORE; lars.eggert@nokia.com
>>>>>> Subject: Re: [sipcore] Call for Consensus: INFO 
>> Specification and 
>>>>>> Congestion
>>>>>>
>>>>>> [as a participant]
>>>>>>
>>>>>> Regarding the proposed text that I sent to the mailing list
>>>>> last week,
>>>>>> there was an exchange regarding where in the document
>>>> this behavior
>>>>>> goes;
>>>>>>
>>>>>> On 5/2/10 6:57 AM, DRAGE, Keith (Keith) wrote:
>>>>>>> ...[T]his text does not belong in section 10, as section 10
>>>>>> is about what needs to be documented to define an INFO
>>>> package, not
>>>>>> about what the implementor actually needs to do.
>>>>>> It actually relates to whether it was appropriate to
>>>> select INFO in
>>>>>> the first place to do the job, as if the information transfer 
>>>>>> requirements are inappropriate to the underlying transport 
>>>>>> capabilities of SIP, then INFO should not have been
>>>> selected in the
>>>>>> first place.
>>>>>>>    
>>>>>> On 5/5/10 8:28 AM, Christer Holmberg responded:
>>>>>>> So, I guess it should be somewhere in section 7.4.
>>>>>>>    
>>>>>> On 5/5/10 8:45 AM, DRAGE, Keith (Keith) replied:
>>>>>>> Yes, but obviously not in the same words. It needs to be
>>>>>> tailored as do not choose an INFO solution if you need 
>> to do this.
>>>>>>>    
>>>>>> I would argue that section 7.3 already goes over this
>>>>> ground in terms
>>>>>> of whether INFO is appropriate for a particular use 
>> case. And, in 
>>>>>> fact,
>>>>>> *most* of the proposed text isn't talking about what
>>>> event package
>>>>>> designers should do. Some of it does, but for the most
>>>>> part, the text
>>>>>> talks about what implementors are required to do. The
>>>>> document *does*
>>>>>> have a section that covers this ground: section 3.
>>>>>>
>>>>>> To that end, I would think we could split this into two
>>>>> sections. The
>>>>>> part dealing with implementation issues would be a new
>>>> section 3.5;
>>>>>> the other would form an introduction to the existing 
>> section 10.9.
>>>>>> Incorporating the other suggestions made so far, then, I
>>>>> would propose
>>>>>> the following sections for the INFO document:
>>>>>>
>>>>>>> 3.5.  Rate and Size of INFO Requests
>>>>>>>
>>>>>>>      INFO messages differ from many other sorts of SIP
>>>> messages in
>>>>>>>      that they carry application information, and the
>>>>> size and rate
>>>>>>>      of the INFO message is directly determined by the
>>>>> application.
>>>>>>>      This can cause application information traffic to interfere
>>>>>>>      with other traffic on that infrastructure, or to
>>>>> self-interfere
>>>>>>>      when data rates become too high.
>>>>>>>
>>>>>>>      Implementors need to consider the impact that the
>>>>> size and the
>>>>>>>      rate of the INFO messages have on the network and on other
>>>>>>>      traffic, since it normally cannot be ensured that
>>>>> INFO messages
>>>>>>>      will be carried over a congestion-controlled
>>>>> transport protocol
>>>>>>>      end-to-end.  Even if an INFO message is sent over such a
>>>>>>>      transport protocol, a downstream SIP entity might
>>>> forward the
>>>>>>>      message over a transport protocol that does not provide
>>>>>>>      congestion control.
>>>>>>>
>>>>>>>      Therefore, an application SHOULD generally have at most one
>>>>>>>      outstanding INFO message.  Implementors are reminded
>>>>> to observe
>>>>>>>      the guidelines in RFC3261 Section 18 regarding the
>>>>> size of SIP
>>>>>>>      messages, such as the guideline to not exceed message sizes
>>>>>>>      of 1300 bytes when the path MTU is unknown. 
>>>> RFC5405 provides
>>>>>>>      additional guidelines for applications using the
>>>>> User Datagram
>>>>>>>      Protocol (UDP) that may be useful background reading.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> 10.9 Rate of INFO Requests
>>>>>>>
>>>>>>>      A designer of an Info package needs to be mindful of the
>>>>>>>      potential congestion impact their application can
>>>>> have on both
>>>>>>>      the SIP network and the Internet at large.  See section 3.5
>>>>>>>      for a discussion of congestion-related issues.
>>>>>>>
>>>>>>>      If there is a maximum or minimum rate at which UAs can send
>>>>>>>      INFO requests associated with the Info Package
>>>>> within a dialog,
>>>>>>>      the Info Package specification MUST document the
>>>> rate values.
>>>>>>>      If the rates can vary, the Info Package specification MAY
>>>>>>>      define Info Package parameters that UAs can use to indicate
>>>>>>>      or negotiate the rates.  Alternatively the rate information
>>>>>>>      can be part of the application data information associated
>>>>>>>      with the Info Package.
>>>>>> Does that sound reasonable?
>>>>>>
>>>>>> /a
>>>>>>
>>> _______________________________________________
>>> sipcore mailing list
>>> sipcore@ietf.org
>>> https://www.ietf.org/mailman/listinfo/sipcore
>>>

From christer.holmberg@ericsson.com  Wed May 19 09:38:32 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: SIPCORE <sipcore@ietf.org>
Date: Wed, 19 May 2010 18:38:12 +0200
Thread-Topic: Keep forward strategy
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Subject: [sipcore] Keep forward strategy
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Hi,

I intend to submit a new version of the keep draft soon.

My intention is to move back to the Via parameter based solution, based on =
comments I've received.

That means that the negotiation will be between adjacent nodes (either duri=
ng initial dialog request, or when the route set has been established).

My intention is also to remove the non-dialog/registration related "ping" u=
se-case, as it seems like OPTIONS would be a better candidate for that. Tha=
t means that keep-alives negotiated during the registration phase will be s=
ent during the lifetime of the registration, and keep-alives negotiated dur=
ing a dialog establishement will be sent during the lifetime of the dialog.

Some other mechanism will then be used for "pings", which lifetime are not =
related to any registration or dialog.

Regards,

Christer

From keith.drage@alcatel-lucent.com  Wed May 19 09:54:13 2010
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
Date: Wed, 19 May 2010 18:52:48 +0200
Thread-Topic: [sipcore] Call for Consensus: INFO Specification and Congestion
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Cc: Adam Roach <adam@estacado.net>, SIPCORE <sipcore@ietf.org>, "lars.eggert@nokia.com" <lars.eggert@nokia.com>
Subject: Re: [sipcore] Call for Consensus: INFO Specification and Congestion
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A compelled signalling mechanism is one where you cannot send the next pack=
et until you have received the ack to the previous packet, i.e. a window si=
ze of one. As such the compelling guarantees the order, and the protocol do=
es not need sequence numbers. I believe the term was in use in telecommunic=
ation signalling systems that behaved this way.

Keith

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: Wednesday, May 19, 2010 2:57 PM
> To: DRAGE, Keith (Keith)
> Cc: Christer Holmberg; Adam Roach; Adam Roach; SIPCORE;
> lars.eggert@nokia.com
> Subject: Re: [sipcore] Call for Consensus: INFO Specification
> and Congestion
>
>
>
> DRAGE, Keith (Keith) wrote:
> > If the transport protocol is used by the application
> protocol in a manner that can get messages out of sequence,
> then the application protocol needs to provide its own
> capabilities to resolve the issue.
>
> In general there is little/nothing the application can do to
> ensure that
>   multiple in-flight INFO messages stay in order. Even if TCP
> is used on every hop, messages may be reordered as they
> transit a proxy. Hence 500 responses due to the out of order
> messages may be received.
>
> And yes, the application protocol can be designed to function
> in the face of that. I'm not suggesting that having multiple
> INFOs outstanding be forbidden - just reminding that there
> are more problems with using them than simply congestion.
>
> > So yes, when using INFO as a compelled signalling
> mechanism, it is reliable.
>
> I don't understand the above. What is a "compelled signalling
> mechanism"?
>
> > If you do not use it that way, then the application has to
> put in its own sequence numbers, or whatever mechanism it so
> desires, to achieve the same purpose. That is of course
> assuming that the application protocol vulnerable to dropped
> or out of sequence messages in the first place.
> >
> > I believe I suggested some time ago on a very early draft
> that we might want to put something about this in the
> document, but no one ever took it up. Too late now.
>
> As long as we are all on the same page about this I'm satisfied.
>
>       Thanks,
>       Paul
>
> > regards
> >
> > Keith
> >
> >> -----Original Message-----
> >> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> >> Sent: Monday, May 17, 2010 3:54 PM
> >> To: DRAGE, Keith (Keith)
> >> Cc: Christer Holmberg; Adam Roach; Adam Roach; SIPCORE;
> >> lars.eggert@nokia.com
> >> Subject: Re: [sipcore] Call for Consensus: INFO Specification and
> >> Congestion
> >>
> >> Maybe this is moot because Keith has agreed to the latest
> proposal...
> >>
> >> I was wondering about your objection to recommending at most one
> >> outstanding INFO message - because there are useful things
> to do with
> >> this.
> >>
> >> In general, having multiple outstanding requests in a dialog is
> >> problematic. With retransmission of lost messages this can
> result in
> >> some messages being rejected due to CSeq errors.
> >> While this isn't necessarily fatal, and there can be a few cases
> >> where this can still have utility, those cases are rare.
> >>
> >>    Thanks,
> >>    Paul
> >>
> >> DRAGE, Keith (Keith) wrote:
> >>> I'm OK with this proposal.
> >>>
> >>> Keith
> >>>
> >>>> -----Original Message-----
> >>>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> >>>> Sent: Monday, May 17, 2010 6:32 AM
> >>>> To: DRAGE, Keith (Keith); Adam Roach
> >>>> Cc: Adam Roach; SIPCORE; lars.eggert@nokia.com
> >>>> Subject: RE: [sipcore] Call for Consensus: INFO
> Specification and
> >>>> Congestion
> >>>>
> >>>>
> >>>> Hi,
> >>>>
> >>>> I appologize that it has taken me a while to reply on
> this issue.
> >>>> I've been busy with 3GPP the last few weeks.
> >>>>
> >>>>
> >>>> Note that the draft already contains section 7.3, which
> >> talks about
> >>>> rate and volume. Nobody has commented on that, so I assume
> >> it is ok.
> >>>> My suggestion would be to:
> >>>>
> >>>> -        Remove section 7.3
> >>>> -        Add new section 7.4.1.2, that talks about rate
> and volume
> >>>> -        New section 7.4.1.2 consists of text from old section
> >>>> 7.3 and the suggested new text for section 10.9.
> >>>> -        Keep section 10.9 unchanged (draft version -07)
> >>>>
> >>>> So, new section 7.4.1.2 could look like:
> >>>>
> >>>>
> >>>> "7.4.1.2.  INFO Request Rate and Volume
> >>>>
> >>>>    INFO messages differ from many other sorts of SIP
> >> messages in that
> >>>>    they carry application information, and the size and
> rate of the
> >>>>    INFO message is directly determined by the
> application. This can
> >>>>    cause application information traffic to interfere with other
> >>>> traffic
> >>>>    on that infrastructure, or to self-interfere when data
> >> rates become
> >>>>    too high.
> >>>>
> >>>>    There is no default throttling mechanism for INFO
> >> requests. Apart
> >>>>    from the SIP session establishment, the number of SIP messages
> >>>>    exchanged during the lifetime a normal SIP session is
> >> rather small.
> >>>>    Some applications, like sending of DTMF tones, can
> >> generate a burst
> >>>>    of up to 20 messages per second.  Other applications,
> >> like constant
> >>>>    GPS location updates, could generate a high rate of
> >> INFO requests
> >>>>    during the lifetime of the invite dialog usage.
> >>>>
> >>>>    A designer of an INFO package, and the application
> that uses it,
> >>>>    need to consider the impact that the size and the rate
> >> of the INFO
> >>>>    messages have on the network and on other traffic, since it
> >>>> normally
> >>>>    cannot be ensured that INFO messages will be carried over a
> >>>>    congestion-controlled transport protocol end-to-end.
> Even if an
> >>>>    INFO message is sent over a such a transport protocol,
> >> a downstream
> >>>>    SIP entity might forward the message over a transport
> >> protocol that
> >>>>    does not provide congestion control.
> >>>>
> >>>>    Furthermore, SIP messages tend to be relatively small,
> >> on the order
> >>>>    of 500 Bytes to 32K Bytes. SIP is a poor mechanism for direct
> >>>>    exchange of bulk data beyond these limits, especially if the
> >>>> headers
> >>>>    plus body exceed the UDP MTU [RFC0768]. Appropriate
> >> mechanisms for
> >>>>    such traffic include HTTP [RFC2616], MSRP [RFC4975], or
> >> other media
> >>>>    plane data transport mechanisms.
> >>>>
> >>>>    RFC5405 provides additional guidelines for applications
> >> using the
> >>>>    User Datagram Protocol (UDP) that may be useful background
> >>>> reading."
> >>>>
> >>>> And, section 10.9 would look like:
> >>>>
> >>>> "10.9.  Rate of INFO Requests
> >>>>
> >>>>    If there is a maximum or minimum rate at which UAs
> can send INFO
> >>>>    requests associated with the Info Package within a
> >> dialog, the Info
> >>>>    Package specification MUST document the rate values.
> >>>>
> >>>>    If the rates can vary, the Info Package specification
> >> MAY define
> >>>> Info
> >>>>    Package parameters that UAs can use to indicate or
> negotiate the
> >>>>    rates.  Alternatively the rate information can be part of the
> >>>>    application data information associated with the Info
> Package."
> >>>>
> >>>>
> >>>> Would people be ok with that?
> >>>>
> >>>> Regards,
> >>>>
> >>>> Christer
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>> -----Original Message-----
> >>>>> From: DRAGE, Keith (Keith)
> [mailto:keith.drage@alcatel-lucent.com]
> >>>>> Sent: 17. toukokuuta 2010 1:03
> >>>>> To: Adam Roach
> >>>>> Cc: Christer Holmberg; Adam Roach; SIPCORE;
> lars.eggert@nokia.com
> >>>>> Subject: RE: [sipcore] Call for Consensus: INFO
> Specification and
> >>>>> Congestion
> >>>>>
> >>>>> This proposed paragraph still causes me issues:
> >>>>>
> >>>>>>      Therefore, an application SHOULD generally have
> at most one
> >>>>>>      outstanding INFO message.  Implementors are reminded
> >>>> to observe
> >>>>>>      the guidelines in RFC3261 Section 18 regarding the
> >>>> size of SIP
> >>>>>>      messages, such as the guideline to not exceed
> message sizes
> >>>>>>      of 1300 bytes when the path MTU is unknown.
> RFC5405 provides
> >>>>>>      additional guidelines for applications using the
> >>>> User Datagram
> >>>>>>      Protocol (UDP) that may be useful background reading.
> >>>>> I already objected to the first sentence above as I do not
> >>>> believe it
> >>>>> solves the issue, and removes useful capabilities that may
> >>>> be desired
> >>>>> by some applications. I CANNOT accept this sentence.
> >>>>>
> >>>>> The second sentence above is an inexact extraction, and as such
> >>>>> misleads completely. The correct quote is below:
> >>>>>
> >>>>>    If a request is within 200 bytes of the path MTU, or
> if it is
> >>>>> larger
> >>>>>    than 1300 bytes and the path MTU is unknown, the
> >> request MUST be
> >>>>> sent
> >>>>>    using an RFC 2914 [43] congestion controlled transport
> >> protocol,
> >>>>> such
> >>>>>    as TCP. If this causes a change in the transport
> >>>> protocol from the
> >>>>>    one indicated in the top Via, the value in the top
> Via MUST be
> >>>>>    changed.  This prevents fragmentation of messages
> over UDP and
> >>>>>    provides congestion control for larger messages.  However,
> >>>>>    implementations MUST be able to handle messages up to
> >> the maximum
> >>>>>    datagram packet size.  For UDP, this size is 65,535 bytes,
> >>>>> including
> >>>>>    IP and UDP headers.
> >>>>>
> >>>>> Nothing about not exceeding message sizes. Only restriction
> >>>> is to send
> >>>>> using a congestion controlled transport protocol if it
> >> exceeds MTU
> >>>>> size. Therefore second sentence as currently drafted is
> >>>> unacceptable.
> >>>>> regards
> >>>>>
> >>>>> Keith
> >>>>>
> >>>>>
> >>>>>> -----Original Message-----
> >>>>>> From: Adam Roach [mailto:adam@nostrum.com]
> >>>>>> Sent: Thursday, May 13, 2010 8:12 PM
> >>>>>> To: DRAGE, Keith (Keith)
> >>>>>> Cc: Christer Holmberg; Adam Roach; SIPCORE;
> lars.eggert@nokia.com
> >>>>>> Subject: Re: [sipcore] Call for Consensus: INFO
> >> Specification and
> >>>>>> Congestion
> >>>>>>
> >>>>>> [as a participant]
> >>>>>>
> >>>>>> Regarding the proposed text that I sent to the mailing list
> >>>>> last week,
> >>>>>> there was an exchange regarding where in the document
> >>>> this behavior
> >>>>>> goes;
> >>>>>>
> >>>>>> On 5/2/10 6:57 AM, DRAGE, Keith (Keith) wrote:
> >>>>>>> ...[T]his text does not belong in section 10, as section 10
> >>>>>> is about what needs to be documented to define an INFO
> >>>> package, not
> >>>>>> about what the implementor actually needs to do.
> >>>>>> It actually relates to whether it was appropriate to
> >>>> select INFO in
> >>>>>> the first place to do the job, as if the information transfer
> >>>>>> requirements are inappropriate to the underlying transport
> >>>>>> capabilities of SIP, then INFO should not have been
> >>>> selected in the
> >>>>>> first place.
> >>>>>>>
> >>>>>> On 5/5/10 8:28 AM, Christer Holmberg responded:
> >>>>>>> So, I guess it should be somewhere in section 7.4.
> >>>>>>>
> >>>>>> On 5/5/10 8:45 AM, DRAGE, Keith (Keith) replied:
> >>>>>>> Yes, but obviously not in the same words. It needs to be
> >>>>>> tailored as do not choose an INFO solution if you need
> >> to do this.
> >>>>>>>
> >>>>>> I would argue that section 7.3 already goes over this
> >>>>> ground in terms
> >>>>>> of whether INFO is appropriate for a particular use
> >> case. And, in
> >>>>>> fact,
> >>>>>> *most* of the proposed text isn't talking about what
> >>>> event package
> >>>>>> designers should do. Some of it does, but for the most
> >>>>> part, the text
> >>>>>> talks about what implementors are required to do. The
> >>>>> document *does*
> >>>>>> have a section that covers this ground: section 3.
> >>>>>>
> >>>>>> To that end, I would think we could split this into two
> >>>>> sections. The
> >>>>>> part dealing with implementation issues would be a new
> >>>> section 3.5;
> >>>>>> the other would form an introduction to the existing
> >> section 10.9.
> >>>>>> Incorporating the other suggestions made so far, then, I
> >>>>> would propose
> >>>>>> the following sections for the INFO document:
> >>>>>>
> >>>>>>> 3.5.  Rate and Size of INFO Requests
> >>>>>>>
> >>>>>>>      INFO messages differ from many other sorts of SIP
> >>>> messages in
> >>>>>>>      that they carry application information, and the
> >>>>> size and rate
> >>>>>>>      of the INFO message is directly determined by the
> >>>>> application.
> >>>>>>>      This can cause application information traffic
> to interfere
> >>>>>>>      with other traffic on that infrastructure, or to
> >>>>> self-interfere
> >>>>>>>      when data rates become too high.
> >>>>>>>
> >>>>>>>      Implementors need to consider the impact that the
> >>>>> size and the
> >>>>>>>      rate of the INFO messages have on the network
> and on other
> >>>>>>>      traffic, since it normally cannot be ensured that
> >>>>> INFO messages
> >>>>>>>      will be carried over a congestion-controlled
> >>>>> transport protocol
> >>>>>>>      end-to-end.  Even if an INFO message is sent over such a
> >>>>>>>      transport protocol, a downstream SIP entity might
> >>>> forward the
> >>>>>>>      message over a transport protocol that does not provide
> >>>>>>>      congestion control.
> >>>>>>>
> >>>>>>>      Therefore, an application SHOULD generally have
> at most one
> >>>>>>>      outstanding INFO message.  Implementors are reminded
> >>>>> to observe
> >>>>>>>      the guidelines in RFC3261 Section 18 regarding the
> >>>>> size of SIP
> >>>>>>>      messages, such as the guideline to not exceed
> message sizes
> >>>>>>>      of 1300 bytes when the path MTU is unknown.
> >>>> RFC5405 provides
> >>>>>>>      additional guidelines for applications using the
> >>>>> User Datagram
> >>>>>>>      Protocol (UDP) that may be useful background reading.
> >>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>>> 10.9 Rate of INFO Requests
> >>>>>>>
> >>>>>>>      A designer of an Info package needs to be mindful of the
> >>>>>>>      potential congestion impact their application can
> >>>>> have on both
> >>>>>>>      the SIP network and the Internet at large.  See
> section 3.5
> >>>>>>>      for a discussion of congestion-related issues.
> >>>>>>>
> >>>>>>>      If there is a maximum or minimum rate at which
> UAs can send
> >>>>>>>      INFO requests associated with the Info Package
> >>>>> within a dialog,
> >>>>>>>      the Info Package specification MUST document the
> >>>> rate values.
> >>>>>>>      If the rates can vary, the Info Package specification MAY
> >>>>>>>      define Info Package parameters that UAs can use
> to indicate
> >>>>>>>      or negotiate the rates.  Alternatively the rate
> information
> >>>>>>>      can be part of the application data information
> associated
> >>>>>>>      with the Info Package.
> >>>>>> Does that sound reasonable?
> >>>>>>
> >>>>>> /a
> >>>>>>
> >>> _______________________________________________
> >>> sipcore mailing list
> >>> sipcore@ietf.org
> >>> https://www.ietf.org/mailman/listinfo/sipcore
> >>>
>

From keith.drage@alcatel-lucent.com  Wed May 19 10:22:01 2010
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To: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>, Paul Kyzivat <pkyzivat@cisco.com>
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Cc: Adam Roach <adam@estacado.net>, SIPCORE <sipcore@ietf.org>, "lars.eggert@nokia.com" <lars.eggert@nokia.com>
Subject: Re: [sipcore] Call for Consensus: INFO Specification and Congestion
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Thinking about it a bit more, maybe the term "compelled" was not the best w=
ord, as it may imply an element of continued or repeated transmission until=
 an ack is received, which is certainly not what I meant to occur at the ap=
plication level.

So I'll stick to the window size of one.

In any case I was just explaining what I meant, and none of this needs a re=
vision of the text.

Keith

> -----Original Message-----
> From: sipcore-bounces@ietf.org
> [mailto:sipcore-bounces@ietf.org] On Behalf Of DRAGE, Keith (Keith)
> Sent: Wednesday, May 19, 2010 5:53 PM
> To: Paul Kyzivat
> Cc: Adam Roach; SIPCORE; lars.eggert@nokia.com
> Subject: Re: [sipcore] Call for Consensus: INFO Specification
> and Congestion
>
> A compelled signalling mechanism is one where you cannot send
> the next packet until you have received the ack to the
> previous packet, i.e. a window size of one. As such the
> compelling guarantees the order, and the protocol does not
> need sequence numbers. I believe the term was in use in
> telecommunication signalling systems that behaved this way.
>
> Keith
>
> > -----Original Message-----
> > From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> > Sent: Wednesday, May 19, 2010 2:57 PM
> > To: DRAGE, Keith (Keith)
> > Cc: Christer Holmberg; Adam Roach; Adam Roach; SIPCORE;
> > lars.eggert@nokia.com
> > Subject: Re: [sipcore] Call for Consensus: INFO Specification and
> > Congestion
> >
> >
> >
> > DRAGE, Keith (Keith) wrote:
> > > If the transport protocol is used by the application
> > protocol in a manner that can get messages out of sequence,
> then the
> > application protocol needs to provide its own capabilities
> to resolve
> > the issue.
> >
> > In general there is little/nothing the application can do to ensure
> > that
> >   multiple in-flight INFO messages stay in order. Even if
> TCP is used
> > on every hop, messages may be reordered as they transit a
> proxy. Hence
> > 500 responses due to the out of order messages may be received.
> >
> > And yes, the application protocol can be designed to
> function in the
> > face of that. I'm not suggesting that having multiple INFOs
> > outstanding be forbidden - just reminding that there are
> more problems
> > with using them than simply congestion.
> >
> > > So yes, when using INFO as a compelled signalling
> > mechanism, it is reliable.
> >
> > I don't understand the above. What is a "compelled signalling
> > mechanism"?
> >
> > > If you do not use it that way, then the application has to
> > put in its own sequence numbers, or whatever mechanism it
> so desires,
> > to achieve the same purpose. That is of course assuming that the
> > application protocol vulnerable to dropped or out of
> sequence messages
> > in the first place.
> > >
> > > I believe I suggested some time ago on a very early draft
> > that we might want to put something about this in the
> document, but no
> > one ever took it up. Too late now.
> >
> > As long as we are all on the same page about this I'm satisfied.
> >
> >       Thanks,
> >       Paul
> >
> > > regards
> > >
> > > Keith
> > >
> > >> -----Original Message-----
> > >> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> > >> Sent: Monday, May 17, 2010 3:54 PM
> > >> To: DRAGE, Keith (Keith)
> > >> Cc: Christer Holmberg; Adam Roach; Adam Roach; SIPCORE;
> > >> lars.eggert@nokia.com
> > >> Subject: Re: [sipcore] Call for Consensus: INFO
> Specification and
> > >> Congestion
> > >>
> > >> Maybe this is moot because Keith has agreed to the latest
> > proposal...
> > >>
> > >> I was wondering about your objection to recommending at most one
> > >> outstanding INFO message - because there are useful things
> > to do with
> > >> this.
> > >>
> > >> In general, having multiple outstanding requests in a dialog is
> > >> problematic. With retransmission of lost messages this can
> > result in
> > >> some messages being rejected due to CSeq errors.
> > >> While this isn't necessarily fatal, and there can be a few cases
> > >> where this can still have utility, those cases are rare.
> > >>
> > >>    Thanks,
> > >>    Paul
> > >>
> > >> DRAGE, Keith (Keith) wrote:
> > >>> I'm OK with this proposal.
> > >>>
> > >>> Keith
> > >>>
> > >>>> -----Original Message-----
> > >>>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> > >>>> Sent: Monday, May 17, 2010 6:32 AM
> > >>>> To: DRAGE, Keith (Keith); Adam Roach
> > >>>> Cc: Adam Roach; SIPCORE; lars.eggert@nokia.com
> > >>>> Subject: RE: [sipcore] Call for Consensus: INFO
> > Specification and
> > >>>> Congestion
> > >>>>
> > >>>>
> > >>>> Hi,
> > >>>>
> > >>>> I appologize that it has taken me a while to reply on
> > this issue.
> > >>>> I've been busy with 3GPP the last few weeks.
> > >>>>
> > >>>>
> > >>>> Note that the draft already contains section 7.3, which
> > >> talks about
> > >>>> rate and volume. Nobody has commented on that, so I assume
> > >> it is ok.
> > >>>> My suggestion would be to:
> > >>>>
> > >>>> -        Remove section 7.3
> > >>>> -        Add new section 7.4.1.2, that talks about rate
> > and volume
> > >>>> -        New section 7.4.1.2 consists of text from old section
> > >>>> 7.3 and the suggested new text for section 10.9.
> > >>>> -        Keep section 10.9 unchanged (draft version -07)
> > >>>>
> > >>>> So, new section 7.4.1.2 could look like:
> > >>>>
> > >>>>
> > >>>> "7.4.1.2.  INFO Request Rate and Volume
> > >>>>
> > >>>>    INFO messages differ from many other sorts of SIP
> > >> messages in that
> > >>>>    they carry application information, and the size and
> > rate of the
> > >>>>    INFO message is directly determined by the
> > application. This can
> > >>>>    cause application information traffic to interfere
> with other
> > >>>> traffic
> > >>>>    on that infrastructure, or to self-interfere when data
> > >> rates become
> > >>>>    too high.
> > >>>>
> > >>>>    There is no default throttling mechanism for INFO
> > >> requests. Apart
> > >>>>    from the SIP session establishment, the number of
> SIP messages
> > >>>>    exchanged during the lifetime a normal SIP session is
> > >> rather small.
> > >>>>    Some applications, like sending of DTMF tones, can
> > >> generate a burst
> > >>>>    of up to 20 messages per second.  Other applications,
> > >> like constant
> > >>>>    GPS location updates, could generate a high rate of
> > >> INFO requests
> > >>>>    during the lifetime of the invite dialog usage.
> > >>>>
> > >>>>    A designer of an INFO package, and the application
> > that uses it,
> > >>>>    need to consider the impact that the size and the rate
> > >> of the INFO
> > >>>>    messages have on the network and on other traffic, since it
> > >>>> normally
> > >>>>    cannot be ensured that INFO messages will be carried over a
> > >>>>    congestion-controlled transport protocol end-to-end.
> > Even if an
> > >>>>    INFO message is sent over a such a transport protocol,
> > >> a downstream
> > >>>>    SIP entity might forward the message over a transport
> > >> protocol that
> > >>>>    does not provide congestion control.
> > >>>>
> > >>>>    Furthermore, SIP messages tend to be relatively small,
> > >> on the order
> > >>>>    of 500 Bytes to 32K Bytes. SIP is a poor mechanism
> for direct
> > >>>>    exchange of bulk data beyond these limits,
> especially if the
> > >>>> headers
> > >>>>    plus body exceed the UDP MTU [RFC0768]. Appropriate
> > >> mechanisms for
> > >>>>    such traffic include HTTP [RFC2616], MSRP [RFC4975], or
> > >> other media
> > >>>>    plane data transport mechanisms.
> > >>>>
> > >>>>    RFC5405 provides additional guidelines for applications
> > >> using the
> > >>>>    User Datagram Protocol (UDP) that may be useful background
> > >>>> reading."
> > >>>>
> > >>>> And, section 10.9 would look like:
> > >>>>
> > >>>> "10.9.  Rate of INFO Requests
> > >>>>
> > >>>>    If there is a maximum or minimum rate at which UAs
> > can send INFO
> > >>>>    requests associated with the Info Package within a
> > >> dialog, the Info
> > >>>>    Package specification MUST document the rate values.
> > >>>>
> > >>>>    If the rates can vary, the Info Package specification
> > >> MAY define
> > >>>> Info
> > >>>>    Package parameters that UAs can use to indicate or
> > negotiate the
> > >>>>    rates.  Alternatively the rate information can be
> part of the
> > >>>>    application data information associated with the Info
> > Package."
> > >>>>
> > >>>>
> > >>>> Would people be ok with that?
> > >>>>
> > >>>> Regards,
> > >>>>
> > >>>> Christer
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>>> -----Original Message-----
> > >>>>> From: DRAGE, Keith (Keith)
> > [mailto:keith.drage@alcatel-lucent.com]
> > >>>>> Sent: 17. toukokuuta 2010 1:03
> > >>>>> To: Adam Roach
> > >>>>> Cc: Christer Holmberg; Adam Roach; SIPCORE;
> > lars.eggert@nokia.com
> > >>>>> Subject: RE: [sipcore] Call for Consensus: INFO
> > Specification and
> > >>>>> Congestion
> > >>>>>
> > >>>>> This proposed paragraph still causes me issues:
> > >>>>>
> > >>>>>>      Therefore, an application SHOULD generally have
> > at most one
> > >>>>>>      outstanding INFO message.  Implementors are reminded
> > >>>> to observe
> > >>>>>>      the guidelines in RFC3261 Section 18 regarding the
> > >>>> size of SIP
> > >>>>>>      messages, such as the guideline to not exceed
> > message sizes
> > >>>>>>      of 1300 bytes when the path MTU is unknown.
> > RFC5405 provides
> > >>>>>>      additional guidelines for applications using the
> > >>>> User Datagram
> > >>>>>>      Protocol (UDP) that may be useful background reading.
> > >>>>> I already objected to the first sentence above as I do not
> > >>>> believe it
> > >>>>> solves the issue, and removes useful capabilities that may
> > >>>> be desired
> > >>>>> by some applications. I CANNOT accept this sentence.
> > >>>>>
> > >>>>> The second sentence above is an inexact extraction,
> and as such
> > >>>>> misleads completely. The correct quote is below:
> > >>>>>
> > >>>>>    If a request is within 200 bytes of the path MTU, or
> > if it is
> > >>>>> larger
> > >>>>>    than 1300 bytes and the path MTU is unknown, the
> > >> request MUST be
> > >>>>> sent
> > >>>>>    using an RFC 2914 [43] congestion controlled transport
> > >> protocol,
> > >>>>> such
> > >>>>>    as TCP. If this causes a change in the transport
> > >>>> protocol from the
> > >>>>>    one indicated in the top Via, the value in the top
> > Via MUST be
> > >>>>>    changed.  This prevents fragmentation of messages
> > over UDP and
> > >>>>>    provides congestion control for larger messages.  However,
> > >>>>>    implementations MUST be able to handle messages up to
> > >> the maximum
> > >>>>>    datagram packet size.  For UDP, this size is 65,535 bytes,
> > >>>>> including
> > >>>>>    IP and UDP headers.
> > >>>>>
> > >>>>> Nothing about not exceeding message sizes. Only restriction
> > >>>> is to send
> > >>>>> using a congestion controlled transport protocol if it
> > >> exceeds MTU
> > >>>>> size. Therefore second sentence as currently drafted is
> > >>>> unacceptable.
> > >>>>> regards
> > >>>>>
> > >>>>> Keith
> > >>>>>
> > >>>>>
> > >>>>>> -----Original Message-----
> > >>>>>> From: Adam Roach [mailto:adam@nostrum.com]
> > >>>>>> Sent: Thursday, May 13, 2010 8:12 PM
> > >>>>>> To: DRAGE, Keith (Keith)
> > >>>>>> Cc: Christer Holmberg; Adam Roach; SIPCORE;
> > lars.eggert@nokia.com
> > >>>>>> Subject: Re: [sipcore] Call for Consensus: INFO
> > >> Specification and
> > >>>>>> Congestion
> > >>>>>>
> > >>>>>> [as a participant]
> > >>>>>>
> > >>>>>> Regarding the proposed text that I sent to the mailing list
> > >>>>> last week,
> > >>>>>> there was an exchange regarding where in the document
> > >>>> this behavior
> > >>>>>> goes;
> > >>>>>>
> > >>>>>> On 5/2/10 6:57 AM, DRAGE, Keith (Keith) wrote:
> > >>>>>>> ...[T]his text does not belong in section 10, as section 10
> > >>>>>> is about what needs to be documented to define an INFO
> > >>>> package, not
> > >>>>>> about what the implementor actually needs to do.
> > >>>>>> It actually relates to whether it was appropriate to
> > >>>> select INFO in
> > >>>>>> the first place to do the job, as if the information
> transfer
> > >>>>>> requirements are inappropriate to the underlying transport
> > >>>>>> capabilities of SIP, then INFO should not have been
> > >>>> selected in the
> > >>>>>> first place.
> > >>>>>>>
> > >>>>>> On 5/5/10 8:28 AM, Christer Holmberg responded:
> > >>>>>>> So, I guess it should be somewhere in section 7.4.
> > >>>>>>>
> > >>>>>> On 5/5/10 8:45 AM, DRAGE, Keith (Keith) replied:
> > >>>>>>> Yes, but obviously not in the same words. It needs to be
> > >>>>>> tailored as do not choose an INFO solution if you need
> > >> to do this.
> > >>>>>>>
> > >>>>>> I would argue that section 7.3 already goes over this
> > >>>>> ground in terms
> > >>>>>> of whether INFO is appropriate for a particular use
> > >> case. And, in
> > >>>>>> fact,
> > >>>>>> *most* of the proposed text isn't talking about what
> > >>>> event package
> > >>>>>> designers should do. Some of it does, but for the most
> > >>>>> part, the text
> > >>>>>> talks about what implementors are required to do. The
> > >>>>> document *does*
> > >>>>>> have a section that covers this ground: section 3.
> > >>>>>>
> > >>>>>> To that end, I would think we could split this into two
> > >>>>> sections. The
> > >>>>>> part dealing with implementation issues would be a new
> > >>>> section 3.5;
> > >>>>>> the other would form an introduction to the existing
> > >> section 10.9.
> > >>>>>> Incorporating the other suggestions made so far, then, I
> > >>>>> would propose
> > >>>>>> the following sections for the INFO document:
> > >>>>>>
> > >>>>>>> 3.5.  Rate and Size of INFO Requests
> > >>>>>>>
> > >>>>>>>      INFO messages differ from many other sorts of SIP
> > >>>> messages in
> > >>>>>>>      that they carry application information, and the
> > >>>>> size and rate
> > >>>>>>>      of the INFO message is directly determined by the
> > >>>>> application.
> > >>>>>>>      This can cause application information traffic
> > to interfere
> > >>>>>>>      with other traffic on that infrastructure, or to
> > >>>>> self-interfere
> > >>>>>>>      when data rates become too high.
> > >>>>>>>
> > >>>>>>>      Implementors need to consider the impact that the
> > >>>>> size and the
> > >>>>>>>      rate of the INFO messages have on the network
> > and on other
> > >>>>>>>      traffic, since it normally cannot be ensured that
> > >>>>> INFO messages
> > >>>>>>>      will be carried over a congestion-controlled
> > >>>>> transport protocol
> > >>>>>>>      end-to-end.  Even if an INFO message is sent
> over such a
> > >>>>>>>      transport protocol, a downstream SIP entity might
> > >>>> forward the
> > >>>>>>>      message over a transport protocol that does not provide
> > >>>>>>>      congestion control.
> > >>>>>>>
> > >>>>>>>      Therefore, an application SHOULD generally have
> > at most one
> > >>>>>>>      outstanding INFO message.  Implementors are reminded
> > >>>>> to observe
> > >>>>>>>      the guidelines in RFC3261 Section 18 regarding the
> > >>>>> size of SIP
> > >>>>>>>      messages, such as the guideline to not exceed
> > message sizes
> > >>>>>>>      of 1300 bytes when the path MTU is unknown.
> > >>>> RFC5405 provides
> > >>>>>>>      additional guidelines for applications using the
> > >>>>> User Datagram
> > >>>>>>>      Protocol (UDP) that may be useful background reading.
> > >>>>>>>
> > >>>>>>>
> > >>>>>>>
> > >>>>>>> 10.9 Rate of INFO Requests
> > >>>>>>>
> > >>>>>>>      A designer of an Info package needs to be
> mindful of the
> > >>>>>>>      potential congestion impact their application can
> > >>>>> have on both
> > >>>>>>>      the SIP network and the Internet at large.  See
> > section 3.5
> > >>>>>>>      for a discussion of congestion-related issues.
> > >>>>>>>
> > >>>>>>>      If there is a maximum or minimum rate at which
> > UAs can send
> > >>>>>>>      INFO requests associated with the Info Package
> > >>>>> within a dialog,
> > >>>>>>>      the Info Package specification MUST document the
> > >>>> rate values.
> > >>>>>>>      If the rates can vary, the Info Package
> specification MAY
> > >>>>>>>      define Info Package parameters that UAs can use
> > to indicate
> > >>>>>>>      or negotiate the rates.  Alternatively the rate
> > information
> > >>>>>>>      can be part of the application data information
> > associated
> > >>>>>>>      with the Info Package.
> > >>>>>> Does that sound reasonable?
> > >>>>>>
> > >>>>>> /a
> > >>>>>>
> > >>> _______________________________________________
> > >>> sipcore mailing list
> > >>> sipcore@ietf.org
> > >>> https://www.ietf.org/mailman/listinfo/sipcore
> > >>>
> >
> _______________________________________________
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From shinji.okumura@softfront.jp  Fri May 21 05:26:35 2010
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From: OKUMURA Shinji <shinji.okumura@softfront.jp>
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Subject: Re: [sipcore] I-D Action:draft-ietf-sipcore-reinvite-04.txt
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Paul,

Paul Kyzivat <pkyzivat@cisco.com>
Wed, 12 May 2010 16:49:31 -0400
>OKUMURA Shinji wrote:
>> Hi,
>> 
>> I have one question and one comment about UAC behavior.
>> 
>> question:
>> At first, oa status is only audio, and during a next reINVITE
>> transaction oa status is changed to audio and video. And then
>> UAS sends a 4xx response to the reINVITE.
>> In this case which is a new offer UAC should send, only audio
>> or both of audio and video with zero port?
>
>Well, the new o/a is to restate the status quo.
>Since the new o/a never succeeded, the status quo includes only one 
>m-line. So I think it would be acceptable to send an offer with that.
>
>OTOH, its possible that the other party is confused and thinks there are 
>two m-lines in effect. It might be further confused to receive another 
>offer with only one m-line. So it might be prudent to send the new offer 
>as you suggest. Its legal to do so, and it ought not hurt anything.
>(But it might cause trouble if the other UA simply can't handle the 
>additional m-line. If so it is quite broken.)

Since both UAs agreed with two m-lines once, probably it does
not cause trouble.

Additionally, IMO a sess-version of SDP should be increased too.

Regards,
Shinji

>> comment:
>> If already-executed changes is made by only PRACK transaction,
>> IMO UAC need not to generate a new offer to synchronize both UAs.
>
>I don't understand what point you are making, or why.
>
>	Thanks,
>	Paul
>
>> Regards,
>> Shinji
>> 
>> Internet-Drafts@ietf.org
>> Sat,  8 May 2010 00:30:02 -0700 (PDT)
>>> 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-04.txt
>>> 	Pages           : 24
>>> 	Date            : 2010-05-08
>>>
>>> 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.
>>>
>>> A URL for this Internet-Draft is:
>>> http://www.ietf.org/internet-drafts/draft-ietf-sipcore-reinvite-04.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.

From brett@broadsoft.com  Fri May 21 12:02:07 2010
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From: Brett Tate <brett@broadsoft.com>
To: "sipcore@ietf.org" <sipcore@ietf.org>
Date: Fri, 21 May 2010 10:34:25 -0700
Thread-Topic: draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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Subject: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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Howdy,

Draft-ietf-sipcore-reinvite-04 sections 3.5 and 4.5 indicate that an UPDATE=
 without SDP should be treated (from a glare perspective) as though it cont=
ained an offer SDP.  However I currently find the definition of "glare" and=
 the expected behavior to be ambiguous.

What is the precise meaning of glare?  I ask mainly because some vendors se=
nd a 491 when an RFC indicates to send a 500.

If INVITE contains offer SDP and 18x was not reliable, how SHOULD the calle=
d party respond if UPDATE without SDP is received from caller: 491, 500, or=
 process as though no glare?

If INVITE contains offer SDP and 18x was not reliable, how SHOULD the calle=
r respond if UPDATE without SDP is received from called party: 491, 500, or=
 process as though no glare?

If caller receives INVITE 2xx without offer/answer, how SHOULD or MUST the =
caller respond if UPDATE without SDP is received from called party prior to=
 sending ACK (if not willing to queue or drop UPDATE): 491 or 500?  Is it a=
 SHOULD or MUST?


Thanks,
Brett


From pkyzivat@cisco.com  Fri May 21 12:53:33 2010
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Subject: Re: [sipcore] I-D Action:draft-ietf-sipcore-reinvite-04.txt
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Shinji,

OKUMURA Shinji wrote:
> Paul,
> 
> Paul Kyzivat <pkyzivat@cisco.com>
> Wed, 12 May 2010 16:49:31 -0400
>> OKUMURA Shinji wrote:
>>> Hi,
>>>
>>> I have one question and one comment about UAC behavior.
>>>
>>> question:
>>> At first, oa status is only audio, and during a next reINVITE
>>> transaction oa status is changed to audio and video. And then
>>> UAS sends a 4xx response to the reINVITE.
>>> In this case which is a new offer UAC should send, only audio
>>> or both of audio and video with zero port?
>> Well, the new o/a is to restate the status quo.
>> Since the new o/a never succeeded, the status quo includes only one 
>> m-line. So I think it would be acceptable to send an offer with that.
>>
>> OTOH, its possible that the other party is confused and thinks there are 
>> two m-lines in effect. It might be further confused to receive another 
>> offer with only one m-line. So it might be prudent to send the new offer 
>> as you suggest. Its legal to do so, and it ought not hurt anything.
>> (But it might cause trouble if the other UA simply can't handle the 
>> additional m-line. If so it is quite broken.)
> 
> Since both UAs agreed with two m-lines once, probably it does
> not cause trouble.

It would be good to explain the reasoning.

> Additionally, IMO a sess-version of SDP should be increased too.

Some problem cases would be the following:

	A                       B
  M1	| reINVITE SDP1,V=a     |
	|---------------------->|
  M2	| 180rel SDP2,V=b       |
	|<----------------------|
  M3	| PRACK SDP3,V=a+1      |
	|---------------------->|
  M4	| 200PRA SDP4,V=b+1     |
	|<----------------------|
  M5	| 4xx INVITE            |
	|<----------------------|
  M6.1	| UPDATE SDP1,V=x       | Either this
	|---------------------->|
  M6.2	| UPDATE SDP4,V=x       | or this
	|---------------------->|

The question above is: should x be 'a', or 'a+2' in M6.*, and does the 
answer to that matter depending on whether the SDP is SDP1 or SDP5?

I think this is moot in most cases because there will be no rejection of 
the invite after two o/a exchanges. But it can happen, say in case of 
preconditions that aren't met.

In M6.1 I think x should be 'a+2'. Does that seem right to you?
I think M6.2 is not recommended. If it occurs, then again I think x 
should be a+2.

	A                       B
  M1	| reINVITE SDP1,V=a     |
	|---------------------->|
  M2	| 180rel SDP2,V=b       |
	|<----------------------|
  M5	| 4xx INVITE            |
	|<----------------------|
  M6.1	| UPDATE SDP1,V=x       | Either this
	|---------------------->|
  M6.2	| UPDATE SDP4,V=x       | or this
	|---------------------->|

In this simpler case, for M6.1 x='a' should be fine and prudent,
because somebody *might* complain if the get two successive SDPs that 
are identical except for the version.

IMO the SDP version is untrustworthy and probably ought to be ignored, 
but it seems that some do check it, depend upon it, and get upset if it 
doesn't behave as they expect.)

	Thanks,
	Paul


> Regards,
> Shinji
> 
>>> comment:
>>> If already-executed changes is made by only PRACK transaction,
>>> IMO UAC need not to generate a new offer to synchronize both UAs.
>> I don't understand what point you are making, or why.
>>
>> 	Thanks,
>> 	Paul
>>
>>> Regards,
>>> Shinji
>>>
>>> Internet-Drafts@ietf.org
>>> Sat,  8 May 2010 00:30:02 -0700 (PDT)
>>>> 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-04.txt
>>>> 	Pages           : 24
>>>> 	Date            : 2010-05-08
>>>>
>>>> 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.
>>>>
>>>> A URL for this Internet-Draft is:
>>>> http://www.ietf.org/internet-drafts/draft-ietf-sipcore-reinvite-04.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.
> 

From shinji.okumura@softfront.jp  Fri May 21 13:38:16 2010
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From: OKUMURA Shinji <shinji.okumura@softfront.jp>
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Subject: Re: [sipcore] I-D Action:draft-ietf-sipcore-reinvite-04.txt
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Gao,

gao.yang2@zte.com.cn
Thu, 13 May 2010 18:08:15 +0800
>Hi Shinji,
>
>UA A MUST finish the first Re-INVITE before the second Re-INVITE. So, 
>there's no ambiguity here. The UA B can handle the Re-INVITE OK.

In 4.5.  Race Conditions and Target Refreshes, 
   Additionally, UAs MUST treat
   the exchange of a 2xx response to an INVITE request and its
   corresponding ACK request as an offer/answer exchange.

What do above descriptions require to UAS?
Should UAS reject a UPDATE received between 2xx response to an
INVITE request and its corresponding ACK or not???

>In my understanding, it is crossing case, while not glare one.

The figure in the previous mail is a glare case of UPDATE.

    A                               B
    |                               |
  s0|re-INV (offer1)                |s0
    |------------------------------>|
    |                   18x(answer1)|
    |<------------------------------|
  s1|                        non-2xx|s1
    |               /---------------|<-+
    |UPD           /                |s0| reject UPDATE
    |=============/================>|* |
    |            /          491(UPD)|  |
    |<==========/===================|  |
    |          /                    |  |
    |<--------/                     |  |
  s0|ACK                 ACK        |  |
    |------------>       ---------->|<-+
    |                               |

And the crossing case of UPDATE is shown as below.

    A                               B
    |                               |
  s0|re-INV (offer1)                |s0
    |------------------------------>|
    |                   18x(answer1)|
    |<------------------------------|
  s1|                        non-2xx|s1
    |<------------------------------|
  s0|                               |s0
    |                   /---------->|*
    |ACK               /  ACK       |
    |---------->      /   --------->|
    |                /              |
    |UPDATE (offer2)/               |
    |--------------/                |
    |                               |

UAS will not be able to recognize the difference between
the glare case and the crossing case.

Regards,
Shinji

>Though it is not normative, but I'd like to distinguish the two:
>
>Glare case: there's ambiguity.
>Crossing case: there's no ambiguity, but message is interlaced from the 
>normal sequence. 
>
>Thanks,
>
>Gao
>
>===================================
> Zip    : 210012
> Tel    : 87211
> Tel2   :(+86)-025-52877211
> e_mail : gao.yang2@zte.com.cn
>===================================
>
>OKUMURA Shinji <shinji.okumura@softfront.jp> 
>2010-05-13 17:35
>
>Gao,
>
>IIUC you means that UAS treats a 4xx to INVITE as an offer and
>an ACK as an answer.
>
>Well, 
>
>    A                               B
>    |                               |
>    |re-INV (offer1)                |
>    |------------------------------>|
>    |                   18x(answer1)|
>    |<------------------------------|
>    |                        non-2xx|
>    |<------------------------------|
>    |                               |
>    |                   /---------->|*
>    |ACK               /            |
>    |-----------------/------------>|
>    |                /              |
>    |re-INV (offer2)/               |
>    |--------------/                |
>    |                               |
>
>In an above Figure, what is a response code to the 2nd
>re-INVITE?
>Or if UA B received the 2nd re-INIVTE before non-2xx,
>what is a response code to it?
>
>gao.yang2@zte.com.cn
>Thu, 13 May 2010 14:00:16 +0800
>>Hi Shinji,
>>
>>There's precondition for the text you mentioned: 
>>
>>When checking for glare situations, UAs MUST treat the exchange of a
>>   non-2xx final response to a re-INVITE and its corresponding ACK
>>   request as an offer/answer exchange.  Consequently, the rules
>>   regarding glare situations applicable to offer/answer exchanges are
>>   also applicable to those exchanges.  These rules imply that if the
>>   UAS of a re-INVITE transaction receives and UPDATE request with an
>>   offer after having sent a non-2xx final response to the re-INVITE but
>>   before having received the corresponding ACK request, the UA SHOULD
>>   return a 491 (Request Pending) response to the UPDATE request. 
>>
>>With the precondition defined here, the UAS(of Re-INVITE) MUST think it 
>>has sended a Offer after it send the non-2xx.
>>
>>Before the UAS get the ACK, it get a new Offer in UPDATE, it SHOULD send 
>>4xx.
>>
>>There's normative definition here.
>
>Is that "normative definition" defined in this draft?
>
>Regards,
>Shinji
>
>
>>Thanks,
>>
>>Gao
>>
>>===================================
>> Zip    : 210012
>> Tel    : 87211
>> Tel2   :(+86)-025-52877211
>> e_mail : gao.yang2@zte.com.cn
>>===================================
>>
>>OKUMURA Shinji <shinji.okumura@softfront.jp> 
>>
>>Hi,
>>
>>In this text,
>>
>>3.5.  Glare Situations
>>   These rules imply that if the
>>   UAS of a re-INVITE transaction receives and UPDATE request with an
>>   offer after having sent a non-2xx final response to the re-INVITE but
>>   before having received the corresponding ACK request, the UA SHOULD
>>   return a 491 (Request Pending) response to the UPDATE request.
>>
>>that is shown in a Figure below.
>>
>>    A                               B
>>    |                               |
>>    |re-INV (offer1)                |
>>    |------------------------------>|
>>    |                   18x(answer1)|
>>    |<------------------------------|
>>    |                        non-2xx|
>>    |               /---------------|<-+
>>    |UPD           /                |  | reject UPDATE
>>    |=============/================>|* |
>>    |            /          491(UPD)|  |
>>    |<==========/===================|  |
>>    |          /                    |  |
>>    |<--------/                     |  |
>>    |ACK                            |  |
>>    |------------------------------>|<-+
>>    |                               |
>>
>>Regarding 491 and 500 response, there are some descriptions in RFC3261
>>and RFC3311. But in these text, an interworking of INVITE and UPDATE
>>is not clear.
>>
>>IMO when the directions of the both request are corresponding,
>>the response code seems to be 500. And if the directions are not
>>corresponding, the response code seems to be 491.
>>
>>Therefore in this case, I think a UAS should return a 500 response to
>>an UPDATE request.
>>
>>Regards,
>>Shinji
>>
>>Internet-Drafts@ietf.org
>>Sat,  8 May 2010 00:30:02 -0700 (PDT)
>>>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-04.txt
>>>                Pages           : 24
>>>                Date            : 2010-05-08
>>>
>>>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.
>>>
>>>A URL for this Internet-Draft is:
>>>http://www.ietf.org/internet-drafts/draft-ietf-sipcore-reinvite-04.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.

From shinji.okumura@softfront.jp  Fri May 21 20:35:08 2010
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Subject: Re: [sipcore] I-D Action:draft-ietf-sipcore-reinvite-04.txt
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Hi all,

In 3.5.  Glare Situations,
|  These rules imply that if the
|  UAS of a re-INVITE transaction receives and UPDATE request with an
|  offer after having sent a non-2xx final response to the re-INVITE but
|  before having received the corresponding ACK request, the UA SHOULD
|  return a 491 (Request Pending) response to the UPDATE request.

There are two variants, shown in Figure 1 and 2 below.

    A                               B
    |                               |
  s0|re-INV (offer1)                |s0
    |------------------------------>|
    |                   18x(answer1)|
    |<------------------------------|
  s1|                        non-2xx|s1
    |               /---------------|<-+
    |UPD(offer2)   /                |s0| reject UPDATE
    |=============/================>|* |
    |            /          491(UPD)|  |
    |<==========/===================|  |
    |          /         ACK        |  |
    |<--------/          ---------->|<-+
  s0|ACK                            |
    |----------->                   |
    |                               |
                Figure 1

    A                               B
    |                               |
  s0|re-INV (offer1)                |s0
    |------------------------------>|
    |                   18x(answer1)|
    |<------------------------------|
  s1|                        non-2xx|s1
    |<------------------------------|<-+
  s0|ACK                            |s0|
    |----------->                   |  |
    |UPD(offer0)                    |  |
    |==============================>|* | reject UPDATE
    |                       491(UPD)|  |
    |<==============================|  |
    |                    ACK        |  |
    |                    ---------->|<-+
    |                               |
                Figure 2

And there is another variant of Figure 1, shown in Figure 3 below.

    A                               B
    |                               |
  s0|re-INV (offer1)                |s0
    |------------------------------>|
    |                   18x(answer1)|
    |<------------------------------|
  s1|                        non-2xx|s1
    |                 /-------------|<-+
    |                /   ACK        |s0| reject UPDATE
    |               /    ---------->|<-+
    |UPD(offer2)   /                |
    |=============/================>|
    |            /          200(UPD)|
    |<==========/===================|
  s2|          /                    |s2
    |<--------/                     |
  s0|ACK                            |
    |----------->                   |
    |                               |
                Figure 3

In Figure 1, even if UAS reject the UPDATE, UAC send new UPDATE
to synchronize a common view in the current rules(Section 3.4).

In Figure 2, if UAS reject the UPDATE, it fails to synchronize
a common view.

In Figure 3, this rule is useless.

After all I think "UAS SHOULD return a 491 between non-2xx and ACK"
is unnecessary.
That is, UAS may return 2xx to this UPDATE.

And, in Figure 1 and 2, the common view of both UAs has already
been synchronized. UAC need not to send an UPDATE request to
synchronize. IMO, If O/A exchange succeeds by the use of UPDATE
before 4xx to INVITE, or UAC has sent UPDATE before 4xx and received
2xx to the UPDATE after 4xx, then UAC should sent new offer to
synchronize.

Regards,
Shinji

Internet-Drafts@ietf.org
Sat,  8 May 2010 00:30:02 -0700 (PDT)
>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-04.txt
>	Pages           : 24
>	Date            : 2010-05-08
>
>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.
>
>A URL for this Internet-Draft is:
>http://www.ietf.org/internet-drafts/draft-ietf-sipcore-reinvite-04.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.

From shinji.okumura@softfront.jp  Fri May 21 22:49:54 2010
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Paul,

>Shinji,
>
>OKUMURA Shinji wrote:
>> Paul,
>> 
>> Paul Kyzivat <pkyzivat@cisco.com>
>> Wed, 12 May 2010 16:49:31 -0400
>>> OKUMURA Shinji wrote:
>>>> Hi,
>>>>
>>>> I have one question and one comment about UAC behavior.
>>>>
>>>> question:
>>>> At first, oa status is only audio, and during a next reINVITE
>>>> transaction oa status is changed to audio and video. And then
>>>> UAS sends a 4xx response to the reINVITE.
>>>> In this case which is a new offer UAC should send, only audio
>>>> or both of audio and video with zero port?
>>> Well, the new o/a is to restate the status quo.
>>> Since the new o/a never succeeded, the status quo includes only one 
>>> m-line. So I think it would be acceptable to send an offer with that.
>>>
>>> OTOH, its possible that the other party is confused and thinks there are 
>>> two m-lines in effect. It might be further confused to receive another 
>>> offer with only one m-line. So it might be prudent to send the new offer 
>>> as you suggest. Its legal to do so, and it ought not hurt anything.
>>> (But it might cause trouble if the other UA simply can't handle the 
>>> additional m-line. If so it is quite broken.)
>> 
>> Since both UAs agreed with two m-lines once, probably it does
>> not cause trouble.
>
>It would be good to explain the reasoning.
>
>> Additionally, IMO a sess-version of SDP should be increased too.
>
>Some problem cases would be the following:
>
>        A                       B
>  M1    | reINVITE SDP1,V=a     |
>        |---------------------->|
>  M2    | 180rel SDP2,V=b       |
>        |<----------------------|
>  M3    | PRACK SDP3,V=a+1      |
>        |---------------------->|
>  M4    | 200PRA SDP4,V=b+1     |
>        |<----------------------|
>  M5    | 4xx INVITE            |
>        |<----------------------|
>  M6.1  | UPDATE SDP1,V=x       | Either this
>        |---------------------->|
>  M6.2  | UPDATE SDP4,V=x       | or this
>        |---------------------->|

Perhaps does you mean M6.2 is "UPDATE SDP5"?

>The question above is: should x be 'a', or 'a+2' in M6.*, and does the 
>answer to that matter depending on whether the SDP is SDP1 or SDP5?

I think x should be a+2, and SDP5 should be based on SDP4.

>I think this is moot in most cases because there will be no rejection of 
>the invite after two o/a exchanges. But it can happen, say in case of 
>preconditions that aren't met.

I agree. this is unclear, but UAC should prepare for it as described
in this draft.

In the following case,

    A                               B
    |                               |
  s0|re-INV (offer1)                |s0
    |------------------------------>|
    |                   18x(answer1)|
    |<------------------------------|
  s1|                        non-2xx|s1
    |                 /-------------|<-+
    |                /   ACK        |s0| reject UPDATE
    |               /    ---------->|<-+
    |UPD(offer2)   /                |
    |=============/================>|
    |            /          200(UPD)|
    |<==========/===================|
  s2|          /                    |s2
    |<--------/                     |
  s0|ACK                            |
    |----------->                   |
    |                               |

s0: only audio
s1: audio and video
s2: audio and video and fax

UA-B recognizes that the current status is s2.
UA-A should send a SDP that is considered reasonable and proper by UA-B.
That is, x is a+2 and m-lines is 3. Certainly video and fax are "zero port".

IMO universal rules should be able to support the most complex cases.

>In M6.1 I think x should be 'a+2'. Does that seem right to you?
>I think M6.2 is not recommended. If it occurs, then again I think x 
>should be a+2.
>
>        A                       B
>  M1    | reINVITE SDP1,V=a     |
>        |---------------------->|
>  M2    | 180rel SDP2,V=b       |
>        |<----------------------|
>  M5    | 4xx INVITE            |
>        |<----------------------|
>  M6.1  | UPDATE SDP1,V=x       | Either this
>        |---------------------->|
>  M6.2  | UPDATE SDP4,V=x       | or this
>        |---------------------->|
>

I assume that PRACK transaction is not shown but exists.

>In this simpler case, for M6.1 x='a' should be fine and prudent,
>because somebody *might* complain if the get two successive SDPs that 
>are identical except for the version.

On the contrary, somebody might complain about a "decrement"
of the sess-version.
At least an increment is not against a rule.

I think x should be a+1 and SDP should be based on SDP2.

Regards,
Shinji

>IMO the SDP version is untrustworthy and probably ought to be ignored, 
>but it seems that some do check it, depend upon it, and get upset if it 
>doesn't behave as they expect.)
>	Thanks,
>	Paul
>
>
>> Regards,
>> Shinji
>> 
>>>> comment:
>>>> If already-executed changes is made by only PRACK transaction,
>>>> IMO UAC need not to generate a new offer to synchronize both UAs.
>>> I don't understand what point you are making, or why.
>>>
>>> 	Thanks,
>>> 	Paul
>>>
>>>> Regards,
>>>> Shinji
>>>>
>>>> Internet-Drafts@ietf.org
>>>> Sat,  8 May 2010 00:30:02 -0700 (PDT)
>>>>> 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-04.txt
>>>>> 	Pages           : 24
>>>>> 	Date            : 2010-05-08
>>>>>
>>>>> 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.
>>>>>
>>>>> A URL for this Internet-Draft is:
>>>>> http://www.ietf.org/internet-drafts/draft-ietf-sipcore-reinvite-04.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.

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From: OKUMURA Shinji <shinji.okumura@softfront.jp>
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Date: Sat, 22 May 2010 17:17:33 +0900
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE withoutSDP
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Hi Brett,

I have a similar question too.

RFC 3261 and 3311 describe the definition regarding 500 and
491 response.
When UA rejects the INVITE during an INVITE transaction, or
rejects the UPDATE during an UPDATE transaction, it has nothing
to do with offer-answer which error code to be used.

If both transaction have the same direction, the error code is 500.
If not, the error code is 491.

But an interworking of INVITE and UPDATE is unclear.
The draft-offeranswer makes an efforts to clarify it in the
section 4.3.

>Howdy,
>
>Draft-ietf-sipcore-reinvite-04 sections 3.5 and 4.5 indicate
>that an UPDATE without SDP should be treated (from a glare
>perspective) as though it contained an offer SDP.
>However I currently find the definition of "glare" and the
>expected behavior to be ambiguous.
>
>What is the precise meaning of glare?  I ask mainly because
>some vendors send a 491 when an RFC indicates to send a 500.
>
>If INVITE contains offer SDP and 18x was not reliable, how
>SHOULD the called party respond if UPDATE without SDP is
>received from caller: 491, 500, or process as though no glare?

If UA must refuse it , parhaps it is 500.
IMO it is another question whether UA must reject the UPDATE
without an offer.

>If INVITE contains offer SDP and 18x was not reliable, how
>SHOULD the caller respond if UPDATE without SDP is received
>from called party: 491, 500, or process as though no glare?

Parhaps it is 491.

>If caller receives INVITE 2xx without offer/answer, how SHOULD
>or MUST the caller respond if UPDATE without SDP is received
>from called party prior to sending ACK (if not willing to
>queue or drop UPDATE): 491 or 500?  Is it a SHOULD or MUST?

UA may send 2xx to UPDATE. Because the INVITE transaction has
been completed.

For reference, the following figures are now under consideration.

  A                      B
  |                      |
  |INVITE                |
  |--------------------->|<-+             <-----+
  |  1xx(offer or answer)|  |                   |
  |<---------------------|  | reject INVITE     | reject UPDATE
  |PRACK                 |  | Don't send INVITE | Don't send UPDATE
  |--------------------->|  |                   |
  |               2xx-PRA|  |                   |
  |<---------------------|  |             <-----+
  |                      |  |                   | may send UPDATE
  |                   2xx|  |                   | may accept UPDATE
  |<---------------------|<-+                   |
  |                      |  | may send INVITE   |
  |ACK                   |  | may accept INVITE |
  |--------------------->|  |                   |
  |                      |  v                   v
  |                      |
          Figure 16

  A                      B
  |                      |
  |INVITE                |
  |--------------------->|<-+             <-----+
  |  1xx(offer or answer)|  |                   |
  |<---------------------|  | reject INVITE     | reject UPDATE
  |PRACK                 |  | Don't send INVITE | Don't send UPDATE
  |--------------------->|  |                   |
  |               2xx-PRA|  |                   |
  |<---------------------|  |             <-----+
  |                      |  |                   | may send UPDATE
  |                   4xx|  |                   | may acceptUPDATE
  |<---------------------|<-+                   |
  |                      |  | Don't send INVITE |
  |ACK          ACK      |  | may accept INVITE | <-- Should B reject UPDATE or not?
  |--------->   -------->|<-+             <-----+     That is another question.
  |                      |  |                   |
  |                      |  v                   v
          Figure 17

In both figures, 
UA A reject all INVITE/UPDATE with 491.
UA B reject all INVITE/UPDATE with 500.

Regards,
Shinji

From brett@broadsoft.com  Sun May 23 10:35:57 2010
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From: Brett Tate <brett@broadsoft.com>
To: OKUMURA Shinji <shinji.okumura@softfront.jp>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Sun, 23 May 2010 10:34:47 -0700
Thread-Topic: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE withoutSDP
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> But an interworking of INVITE and UPDATE is unclear.
> The draft-offeranswer makes an efforts to clarify it in the
> section 4.3.

Thanks for the information.  Draft-ietf-sipping-sip-offeranswer needs to be=
 updated to actually refer to draft-ietf-sipcore-reinvite.


> >If INVITE contains offer SDP and 18x was not reliable, how
> >SHOULD the called party respond if UPDATE without SDP is
> >received from caller: 491, 500, or process as though no glare?
>=20
> If UA must refuse it , perhaps it is 500.

If I understand draft-ietf-sipcore-reinvite and RFC3311 correctly, the foll=
owing snippets apply.

"When checking for glare situations, UAs MUST treat every UPDATE request as=
 if it contained an offer."

"Similarly, if an UPDATE is received that contains an offer, and the UAS ha=
s received an offer (in an UPDATE, PRACK, or INVITE) to which it has not ye=
t generated an answer, the UAS MUST reject the UPDATE with a 500 response, =
and MUST include a Retry-After header field with a randomly chosen value be=
tween 0 and 10 seconds."


> IMO it is another question whether UA must reject the UPDATE
> without an offer.

My current understanding is that it MUST be rejected; see above.


> >If INVITE contains offer SDP and 18x was not reliable, how
> >SHOULD the caller respond if UPDATE without SDP is received
> >from called party: 491, 500, or process as though no glare?
>=20
> Perhaps it is 491.

If I understand draft-ietf-sipcore-reinvite and RFC3311 correctly, the foll=
owing snippets apply.

"When checking for glare situations, UAs MUST treat every UPDATE request as=
 if it contained an offer."

"If an UPDATE is received that contains an offer, and the UAS has generated=
 an offer (in an UPDATE, PRACK or INVITE) to which it has not yet received =
an answer, the UAS MUST reject the UPDATE with a 491 response."

If the calling party supports draft-ietf-sipcore-reinvite but called party =
doesn't, this can lead to numerous UPDATE/491 (retry interval 0-2 seconds) =
until called party no longer mandated to return 491.


> >If caller receives INVITE 2xx without offer/answer, how SHOULD
> >or MUST the caller respond if UPDATE without SDP is received
> >from called party prior to sending ACK (if not willing to
> >queue or drop UPDATE): 491 or 500?  Is it a SHOULD or MUST?
>=20
> UA may send 2xx to UPDATE. Because the INVITE transaction has
> been completed.

If I understand draft-ietf-sipcore-reinvite and RFC3311 correctly, the foll=
owing snippets apply.

"When checking for glare situations, UAs MUST treat every UPDATE request as=
 if it contained an offer.  Additionally, UAs MUST treat the exchange of a =
2xx response to an INVITE request and its corresponding ACK request as an o=
ffer/answer exchange."

"Similarly, if an UPDATE is received that contains an offer, and the UAS ha=
s received an offer (in an UPDATE, PRACK, or INVITE) to which it has not ye=
t generated an answer, the UAS MUST reject the UPDATE with a 500 response, =
and MUST include a Retry-After header field with a randomly chosen value be=
tween 0 and 10 seconds."


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Date: Mon, 24 May 2010 11:10:43 +0530
From: Harbhanu <harbhanu@huawei.com>
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Subject: [sipcore] REg. draft-ietf-sipcore-reinvite-04
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1. UE1 --- INVITE SE-1800 --> UE2

2. UE1 <--- 183 ------------------ UE2

 

3. UE1 --- UPDATE SE-600 --> UE2

4. UE1 <--- 200 UPDATE -SE600 -- UE2

 

5. UE1 <--- 200 INVITE SE-1800 -- UE2

 

Please refer the scenario above for handling of Session-Timer, (4028). 

As per the Draft-ietf-sipcore-reinvite-04 section 4.5:

"  When checking for glare situations, UAs MUST treat every UPDATE

   request as if it contained an offer.  Additionally, UAs MUST treat

   the exchange of a 2xx response to an INVITE request and its

   corresponding ACK request as an offer/answer exchange.  Consequently,

   the rules regarding glare situations applicable to offer/answer

   exchanges are also applicable to those exchanges."

 

Here, we are not able to conclude whether both UAs should run session timer
with 

1800 or 600, or otherwise whether the UPDATE should be rejected.

 

As per my understanding this draft doesn't *explicitly* mention anything
about the Session-Timer, wherein there is an early negotiation. Since 4028
also doesn't 

have any reference for early negotiation, so we suggest that it can better
be incorporated in the above mentioned draft[ietf-sipcore-reinvite-04].

 

Please share your opinion and do guide us incase any of the existing
RFC/active-draft already captures this.

 

Thanks!

 

Regards,

Harbhanu

 

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>1. UE1 --- INVITE SE-1800 </span></font><font =
size=3D2
face=3DWingdings><span =
style=3D'font-size:10.0pt;font-family:Wingdings'>&agrave;</span></font><f=
ont
size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'> =
UE2<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>2. UE1 &lt;--- 183 ------------------ =
UE2<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
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<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>3. UE1 --- UPDATE SE-600 </span></font><font size=3D2
face=3DWingdings><span =
style=3D'font-size:10.0pt;font-family:Wingdings'>&agrave;</span></font><f=
ont
size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'> =
UE2<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>4. UE1 &lt;--- 200 UPDATE -SE600 -- =
UE2<o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>5. UE1 &lt;--- 200 INVITE SE-1800 -- =
UE2<o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Please refer the scenario above for handling of =
Session-Timer,
(4028). <o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3D"#1f497d" =
face=3DCalibri><span
style=3D'font-size:11.0pt;font-family:Calibri;color:#1F497D'>As per the =
Draft-ietf-sipcore-reinvite-04
section 4.5:<o:p></o:p></span></font></p>

<pre style=3D'page-break-before:always'><font size=3D2 color=3D"#1f497d"
face=3DCalibri><span =
style=3D'font-size:11.0pt;font-family:Calibri;color:#1F497D'>&#8220;</spa=
n></font><span
lang=3DEN>&nbsp; When checking for glare situations, UAs MUST treat =
every UPDATE</span><font
face=3D"Courier New"><span lang=3DEN style=3D'font-family:"Courier =
New"'><o:p></o:p></span></font></pre>

<p class=3DMsoNormal style=3D'page-break-before:always'><font size=3D2
face=3D"Courier New"><span lang=3DEN =
style=3D'font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;
request as if it contained an offer.&nbsp; Additionally, UAs MUST =
treat<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'page-break-before:always'><font size=3D2
face=3D"Courier New"><span lang=3DEN =
style=3D'font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;
the exchange of a 2xx response to an INVITE request and =
its<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'page-break-before:always'><font size=3D2
face=3D"Courier New"><span lang=3DEN =
style=3D'font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;
corresponding ACK request as an offer/answer exchange.&nbsp; =
Consequently,<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'page-break-before:always'><font size=3D2
face=3D"Courier New"><span lang=3DEN =
style=3D'font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;
the rules regarding glare situations applicable to =
offer/answer<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'page-break-before:always'><font size=3D2
face=3D"Courier New"><span lang=3DEN =
style=3D'font-size:10.0pt;font-family:"Courier New"'>&nbsp;&nbsp;
exchanges are also applicable to those =
exchanges.&#8221;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3D"#1f497d" =
face=3DCalibri><span
style=3D'font-size:11.0pt;font-family:Calibri;color:#1F497D'><o:p>&nbsp;<=
/o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Here, we are not able to conclude whether both UAs =
should run
session timer with <o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>1800 or 600, or otherwise whether the UPDATE should =
be
rejected.<o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>As per my understanding this draft doesn&#8217;t =
*<b><span
style=3D'font-weight:bold'>explicitly</span></b>* mention anything about =
the
Session-Timer, wherein there is an early negotiation. Since 4028 also =
doesn&#8217;t
<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>have any reference for early negotiation, so we =
suggest that
it can better be incorporated in the above mentioned =
draft[</span></font><font
size=3D2 color=3D"#1f497d" face=3DCalibri><span =
style=3D'font-size:11.0pt;font-family:
Calibri;color:#1F497D'>ietf-sipcore-reinvite-04</span></font><font =
size=3D2
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>].<o:p></o:p></span></font><=
/p>

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Please share your opinion and do guide us incase any =
of the
existing RFC/active-draft already captures =
this.<o:p></o:p></span></font></p>

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

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

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

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

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

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>******************************************************=
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Date: Mon, 24 May 2010 12:54:00 +0530
From: Harbhanu <harbhanu@huawei.com>
In-reply-to: <OFAD8B7FED.AAEB71E6-ON4825772D.0023D33D-4825772D.0025D4A9@zte.com.cn>
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Cc: sipcore@ietf.org
Subject: Re: [sipcore] REg. draft-ietf-sipcore-reinvite-04  //Session Timer
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Honest thanks for a prompt reply.. That said, below are some =
clarifications
for your mail -

=20

>> Do it like target refresh, then the 2xx of INVITE/Re-INVITE's SE =
value
should be as the SE value in the last embedded subsequent session =
refresh
request's 2xx.=20



But then it might break the existing clause mentioned in RFC-4028, when =
the
scenario has INV(SE-600) & UPDATE(SE-1800),that =A8C
=A1=B0 The UAS MUST NOT increase the value of the Session-Expires header
field.=A1=B1

=20

>> So, I think it is not recommended to sending UPDATE as subsequent =
session
refresh request in pending INVITE/Re-INVITE.

I suggest to incorporate its clear handling in this draft itself, since
anyways it makes an excellent attempt to clear the muck around all the
refresh-request handling.

IMO this will make the draft a bit more comprehensive and might
baseline/standardize the behavior for other extension handlings.

=20

Thanks again!

=20

Regards,

Harbhanu

=20

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  _____ =20

From: gao.yang2@zte.com.cn [mailto:gao.yang2@zte.com.cn]=20
Sent: Monday, May 24, 2010 12:21 PM
To: Harbhanu
Cc: sipcore@ietf.org
Subject: Re: [sipcore] REg. draft-ietf-sipcore-reinvite-04 //Session =
Timer

=20


Hi,=20

Not like O/A usage and target refresh usage, there's no explicit =
definition
of allowing subsequent session refresh request in embedded requests(such =
as
UPDATE during pending Re-INVITE). So, I think it is not recommended to
sending UPDATE as subsequent session refresh request in pending
INVITE/Re-INVITE.=20

I'd like to talk more about session timer. If people want to do the
semantics extension for session timer's to allow embedded requests(such =
as
UPDATE during pending Re-INVITE), there's one suggestion way(INFORMATIVE =
at
all):=20

Do it like target refresh, then the 2xx of INVITE/Re-INVITE's SE value
should be as the SE value in the last embedded subsequent session =
refresh
request's 2xx.=20

Then, the glare handling for such usage would be the same as the Target
Refresh part in draft-ietf-sipcore-reinvite.=20

Thanks,=20

Gao=20

=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
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Zip    : 210012
Tel    : 87211
Tel2   :(+86)-025-52877211
e_mail : gao.yang2@zte.com.cn
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=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=20




Harbhanu <harbhanu@huawei.com>=20
=B7=A2=BC=FE=C8=CB:  sipcore-bounces@ietf.org=20

2010-05-24 13:40=20


=CA=D5=BC=FE=C8=CB

sipcore@ietf.org=20


=B3=AD=CB=CD

=20


=D6=F7=CC=E2

[sipcore] REg. draft-ietf-sipcore-reinvite-04

=20


=20

=20




 =20
1. UE1 --- INVITE SE-1800 --> UE2=20
2. UE1 <--- 183 ------------------ UE2=20
 =20
3. UE1 --- UPDATE SE-600 --> UE2=20
4. UE1 <--- 200 UPDATE -SE600 -- UE2=20
 =20
5. UE1 <--- 200 INVITE SE-1800 -- UE2=20
 =20
Please refer the scenario above for handling of Session-Timer, (4028).=20
As per the Draft-ietf-sipcore-reinvite-04 section 4.5:=20
=A1=B0  When checking for glare situations, UAs MUST treat every UPDATE=20
   request as if it contained an offer.  Additionally, UAs MUST treat=20
   the exchange of a 2xx response to an INVITE request and its=20
   corresponding ACK request as an offer/answer exchange.  Consequently, =

   the rules regarding glare situations applicable to offer/answer=20
   exchanges are also applicable to those exchanges.=A1=B1=20
 =20
Here, we are not able to conclude whether both UAs should run session =
timer
with=20
1800 or 600, or otherwise whether the UPDATE should be rejected.=20
 =20
As per my understanding this draft doesn=A1=AFt *explicitly* mention =
anything
about the Session-Timer, wherein there is an early negotiation. Since =
4028
also doesn=A1=AFt=20
have any reference for early negotiation, so we suggest that it can =
better
be incorporated in the above mentioned draft[ietf-sipcore-reinvite-04].=20
 =20
Please share your opinion and do guide us incase any of the existing
RFC/active-draft already captures this.=20
 =20
Thanks!=20
 =20
Regards,=20
Harbhanu=20
 =20
*************************************************************************=
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<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Honest thanks for a prompt reply.. =
That
said, below are some clarifications for your mail =
-<o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>&gt;&gt;</span></font><font =
size=3D2
face=3Dsans-serif><span =
style=3D'font-size:10.0pt;font-family:sans-serif'> Do it
like target refresh, then the 2xx of INVITE/Re-INVITE's SE value should =
be as
the SE value in the last embedded subsequent session refresh request's =
2xx.</span></font>
<br>
<br>
<font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:10.0pt;font-family:
Arial;color:navy'><o:p></o:p></span></font></p>

<pre><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial;color:navy'>But then it might break the existing =
clause mentioned in RFC-4028, when the scenario has INV(SE-600) &amp; =
UPDATE(SE-1800),that =A8C<o:p></o:p></span></font></pre><pre><font
size=3D3 color=3Dnavy face=3DArial><span =
style=3D'font-size:12.0pt;font-family:Arial;
color:navy'>=A1=B0 </span></font>The UAS MUST NOT increase the value of =
the Session-Expires header field.<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>=A1=B1</span></font><o:p></o:p></pre>

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>&gt;&gt; </span></font><font =
size=3D2
face=3Dsans-serif><span =
style=3D'font-size:10.0pt;font-family:sans-serif'>So, I
think it is not recommended to sending UPDATE as subsequent session =
refresh
request in pending INVITE/Re-INVITE.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>I suggest to incorporate its clear
handling in this draft itself, since anyways it makes an excellent =
attempt to
clear the muck around all the refresh-request =
handling.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>IMO this will make the draft a bit =
more
comprehensive and might baseline/standardize the behavior for other =
extension
handlings.<o:p></o:p></span></font></p>

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

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

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

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

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

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>************************************=
***************************************************<br>
This e-mail and attachments contain confidential information from =
HUAWEI, which
is intended only for the person or entity whose address is listed above. =
Any
use of the information contained herein in any way (including, but not =
limited
to, total or partial disclosure, reproduction, or dissemination) by =
persons
other than the intended recipient's) is prohibited. If you receive this =
e-mail
in error, please notify the sender by phone or email immediately and =
delete it!</span></font><o:p></o:p></p>

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<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font =
size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> =
gao.yang2@zte.com.cn
[mailto:gao.yang2@zte.com.cn] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Monday, May 24, =
2010 12:21
PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Harbhanu<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> sipcore@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: [sipcore] =
REg.
draft-ietf-sipcore-reinvite-04 //Session =
Timer</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 face=3DSimSun><span =
style=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></span></font></p>

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6!!!!!!!!!!1113308@D@E@EOnsl`m/enu!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!=
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!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!1!1"=20
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style=3D'position:absolute;margin-left:0;margin-top:0;width:.05pt;height:=
.05pt;
 z-index:1;visibility:hidden'>
 <w:anchorlock/>
</v:shape><![endif]--></span><br>
</font><font size=3D2 face=3Dsans-serif><span =
style=3D'font-size:10.0pt;font-family:
sans-serif'>Hi,</span></font> <br>
<br>
<font size=3D2 face=3Dsans-serif><span =
style=3D'font-size:10.0pt;font-family:sans-serif'>Not
like O/A usage and target refresh usage, there's no explicit definition =
of
allowing subsequent session refresh request in embedded requests(such as =
UPDATE
during pending Re-INVITE). So, I think it is not recommended to sending =
UPDATE
as subsequent session refresh request in pending =
INVITE/Re-INVITE.</span></font>
<br>
<br>
<font size=3D2 face=3Dsans-serif><span =
style=3D'font-size:10.0pt;font-family:sans-serif'>I'd
like to talk more about session timer. If people want to do the =
semantics
extension for session timer's to allow embedded requests(such as UPDATE =
during
pending Re-INVITE), there's one suggestion way(INFORMATIVE at =
all):</span></font>
<br>
<br>
<font size=3D2 face=3Dsans-serif><span =
style=3D'font-size:10.0pt;font-family:sans-serif'>Do
it like target refresh, then the 2xx of INVITE/Re-INVITE's SE value =
should be
as the SE value in the last embedded subsequent session refresh =
request's 2xx.</span></font>
<br>
<br>
<font size=3D2 face=3Dsans-serif><span =
style=3D'font-size:10.0pt;font-family:sans-serif'>Then,
the glare handling for such usage would be the same as the Target =
Refresh part
in draft-ietf-sipcore-reinvite.</span></font> <br>
<br>
<font size=3D2 face=3Dsans-serif><span =
style=3D'font-size:10.0pt;font-family:sans-serif'>Thanks,</span></font>
<br>
<br>
<font size=3D2 face=3Dsans-serif><span =
style=3D'font-size:10.0pt;font-family:sans-serif'>Gao</span></font>
<br>
<br>
<font size=3D2 face=3Dsans-serif><span =
style=3D'font-size:10.0pt;font-family:sans-serif'>=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D<br>
Zip &nbsp; &nbsp;: 210012<br>
Tel &nbsp; &nbsp;: 87211<br>
Tel2 &nbsp; :(+86)-025-52877211<br>
e_mail : gao.yang2@zte.com.cn<br>
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D</span></font> <br>
<br>
<o:p></o:p></p>

<table class=3DMsoNormalTable border=3D0 cellpadding=3D0 width=3D"100%"
 style=3D'width:100.0%'>
 <tr>
  <td width=3D"26%" valign=3Dtop style=3D'width:26.0%;padding:.75pt =
.75pt .75pt .75pt'>
  <p class=3DMsoNormal><b><font size=3D1 face=3Dsans-serif><span =
style=3D'font-size:
  7.5pt;font-family:sans-serif;font-weight:bold'>Harbhanu
  &lt;harbhanu@huawei.com&gt;</span></font></b><font size=3D1 =
face=3Dsans-serif><span
  style=3D'font-size:7.5pt;font-family:sans-serif'> </span></font><br>
  <font size=3D1><span lang=3DZH-CN =
style=3D'font-size:7.5pt'>=B7=A2=BC=FE=C8=CB</span></font><font
  size=3D1 face=3Dsans-serif><span =
style=3D'font-size:7.5pt;font-family:sans-serif'>:
  &nbsp;sipcore-bounces@ietf.org</span></font> <o:p></o:p></p>
  <p><font size=3D1 face=3Dsans-serif><span =
style=3D'font-size:7.5pt;font-family:
  sans-serif'>2010-05-24 13:40</span></font> <o:p></o:p></p>
  </td>
  <td width=3D"73%" valign=3Dtop style=3D'width:73.0%;padding:.75pt =
.75pt .75pt .75pt'>
  <table class=3DMsoNormalTable border=3D0 cellpadding=3D0 =
width=3D"100%"
   style=3D'width:100.0%'>
   <tr>
    <td valign=3Dtop style=3D'padding:.75pt .75pt .75pt .75pt'>
    <p class=3DMsoNormal align=3Dright style=3D'text-align:right'><font =
size=3D1
    face=3DSimSun><span lang=3DZH-CN =
style=3D'font-size:7.5pt'>=CA=D5=BC=FE=C8=CB</span></font><o:p></o:p></p>=

    </td>
    <td valign=3Dtop style=3D'padding:.75pt .75pt .75pt .75pt'>
    <p class=3DMsoNormal><font size=3D1 face=3Dsans-serif><span =
style=3D'font-size:
    7.5pt;font-family:sans-serif'>sipcore@ietf.org</span></font> =
<o:p></o:p></p>
    </td>
   </tr>
   <tr>
    <td valign=3Dtop style=3D'padding:.75pt .75pt .75pt .75pt'>
    <p class=3DMsoNormal align=3Dright style=3D'text-align:right'><font =
size=3D1
    face=3DSimSun><span lang=3DZH-CN =
style=3D'font-size:7.5pt'>=B3=AD=CB=CD</span></font><o:p></o:p></p>
    </td>
    <td valign=3Dtop style=3D'padding:.75pt .75pt .75pt .75pt'>
    <p class=3DMsoNormal><font size=3D3 face=3DSimSun><span =
style=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></span></font></p>
    </td>
   </tr>
   <tr>
    <td valign=3Dtop style=3D'padding:.75pt .75pt .75pt .75pt'>
    <p class=3DMsoNormal align=3Dright style=3D'text-align:right'><font =
size=3D1
    face=3DSimSun><span lang=3DZH-CN =
style=3D'font-size:7.5pt'>=D6=F7=CC=E2</span></font><o:p></o:p></p>
    </td>
    <td valign=3Dtop style=3D'padding:.75pt .75pt .75pt .75pt'>
    <p class=3DMsoNormal><font size=3D1 face=3Dsans-serif><span =
style=3D'font-size:
    7.5pt;font-family:sans-serif'>[sipcore] REg. =
draft-ietf-sipcore-reinvite-04</span></font><o:p></o:p></p>
    </td>
   </tr>
  </table>
  <p class=3DMsoNormal><font size=3D3 face=3DSimSun><span =
style=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></span></font></p>
  <table class=3DMsoNormalTable border=3D0 cellpadding=3D0>
   <tr>
    <td valign=3Dtop style=3D'padding:.75pt .75pt .75pt .75pt'>
    <p class=3DMsoNormal><font size=3D3 face=3DSimSun><span =
style=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></span></font></p>
    </td>
    <td valign=3Dtop style=3D'padding:.75pt .75pt .75pt .75pt'>
    <p class=3DMsoNormal><font size=3D3 face=3DSimSun><span =
style=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></span></font></p>
    </td>
   </tr>
  </table>
  <p class=3DMsoNormal><font size=3D3 face=3DSimSun><span =
style=3D'font-size:12.0pt'><o:p></o:p></span></font></p>
  </td>
 </tr>
</table>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D3 =
face=3DSimSun><span
style=3D'font-size:12.0pt'><br>
<br>
<br>
</span></font><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:
Arial'>&nbsp;</span></font> <br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>1. UE1
--- INVITE SE-1800 </span></font><font size=3D2 face=3DWingdings><span
style=3D'font-size:10.0pt;font-family:Wingdings'>=A8=A4</span></font><fon=
t size=3D2
face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'> =
UE2</span></font> <br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>2. UE1
&lt;--- 183 ------------------ UE2</span></font> <br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>&nbsp;</span></font>
<br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>3. UE1
--- UPDATE SE-600 </span></font><font size=3D2 face=3DWingdings><span
style=3D'font-size:10.0pt;font-family:Wingdings'>=A8=A4</span></font><fon=
t size=3D2
face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'> =
UE2</span></font> <br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>4. UE1
&lt;--- 200 UPDATE -SE600 -- UE2</span></font> <br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>&nbsp;</span></font>
<br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>5. UE1
&lt;--- 200 INVITE SE-1800 -- UE2</span></font> <br>
<font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:10.0pt;font-family:
Arial;color:navy'>&nbsp;</span></font> <br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>Please
refer the scenario above for handling of Session-Timer, (4028). =
</span></font><br>
<font size=3D2 color=3D"#004080"><span =
style=3D'font-size:10.0pt;color:#004080'>As
per the Draft-ietf-sipcore-reinvite-04 section 4.5:</span></font> <br>
<font size=3D2 color=3D"#004080" face=3D"Times New Roman"><span =
style=3D'font-size:
10.0pt;font-family:"Times New =
Roman";color:#004080'>=A1=B0</span></font><font
size=3D2><span style=3D'font-size:10.0pt'> </span></font><font size=3D2
face=3D"Times New Roman"><span =
style=3D'font-size:10.0pt;font-family:"Times New =
Roman"'>&nbsp;</span></font><font
size=3D2><span style=3D'font-size:10.0pt'>When checking for glare =
situations, UAs
MUST treat every UPDATE</span></font> <br>
<font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>&nbsp;
&nbsp;request as if it contained an offer. &nbsp;Additionally, UAs MUST =
treat</span></font>
<br>
<font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>&nbsp;
&nbsp;the exchange of a 2xx response to an INVITE request and =
its</span></font>
<br>
<font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>&nbsp;
&nbsp;corresponding ACK request as an offer/answer exchange.
&nbsp;Consequently,</span></font> <br>
<font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>&nbsp;
&nbsp;the rules regarding glare situations applicable to =
offer/answer</span></font>
<br>
<font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>&nbsp;
&nbsp;exchanges are also applicable to those =
exchanges.=A1=B1</span></font> <br>
<font size=3D2 color=3D"#004080" face=3D"Times New Roman"><span =
style=3D'font-size:
10.0pt;font-family:"Times New Roman";color:#004080'>&nbsp;</span></font> =
<br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>Here,
we are not able to conclude whether both UAs should run session timer =
with </span></font><br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>1800
or 600, or otherwise whether the UPDATE should be =
rejected.</span></font> <br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>&nbsp;</span></font>
<br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>As per
my understanding this draft doesn=A1=AFt *<b><span =
style=3D'font-weight:bold'>explicitly</span></b>*
mention anything about the Session-Timer, wherein there is an early =
negotiation.
Since 4028 also doesn=A1=AFt </span></font><br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>have
any reference for early negotiation, so we suggest that it can better be
incorporated in the above mentioned draft[</span></font><font size=3D2
color=3D"#004080"><span =
style=3D'font-size:10.0pt;color:#004080'>ietf-sipcore-reinvite-04</span><=
/font><font
size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>].</span></font>
<br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>&nbsp;</span></font>
<br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>Please
share your opinion and do guide us incase any of the existing =
RFC/active-draft
already captures this.</span></font> <br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>&nbsp;</span></font>
<br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>Thanks!</span></font>
<br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>&nbsp;</span></font>
<br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>Regards,</span></font>
<br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>Harbhanu</span></font>
<br>
<font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>&nbsp;</span></font>
<br>
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<pre><font size=3D3 face=3DSimSun><span =
style=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><fon=
t
size=3D3 face=3DSimSun><span =
style=3D'font-size:12.0pt'>----------------------------------------------=
----------<o:p></o:p></span></font></pre><pre><font
size=3D3 face=3DSimSun><span =
style=3D'font-size:12.0pt'>ZTE</span></font><font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>Information<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>Security<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>Notice:<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>The<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>information<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>contained<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>in<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>this<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>mail<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>is<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>solely<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>property<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>of<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>the<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>sender's<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>organization.<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>This<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>mail<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>communication<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>is<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>confidential.<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>Recipients<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>named<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>above<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>are<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>obligated<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>to<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>maintain<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>secrecy<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>and<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>are<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>not<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>permitted<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>to<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>disclose<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>the<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>contents<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>of<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>this<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>communication<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>to<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>others.<o:p></o:p></pre><pre><font
size=3D3 face=3DSimSun><span =
style=3D'font-size:12.0pt'>This</span></font><font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>email<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>and<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>any<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>files<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>transmitted<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>with<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>it<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>are<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>confidential<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>and<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>intended<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>solely<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>for<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>the<font
face=3D"Courier New"><span style=3D'font-family:"Courier =
New"'>&nbsp;</span></font>use<font
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New"'>&nbsp;</span></font>system.<o:p></o:p></pre></div>

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From shinji.okumura@softfront.jp  Mon May 24 01:02:43 2010
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From: OKUMURA Shinji <shinji.okumura@softfront.jp>
To: "sipcore@ietf.org" <sipcore@ietf.org>
Date: Mon, 24 May 2010 17:02:30 +0900
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE withoutSDP
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Brett,

Brett Tate <brett@broadsoft.com>
Sun, 23 May 2010 10:34:47 -0700
(snip)
>> >If caller receives INVITE 2xx without offer/answer, how SHOULD
>> >or MUST the caller respond if UPDATE without SDP is received
>> >from called party prior to sending ACK (if not willing to
>> >queue or drop UPDATE): 491 or 500?  Is it a SHOULD or MUST?
>> 
>> UA may send 2xx to UPDATE. Because the INVITE transaction has
>> been completed.
>
>If I understand draft-ietf-sipcore-reinvite and RFC3311 correctly,
>the following snippets apply.
>
>"When checking for glare situations, UAs MUST treat every UPDATE
>request as if it contained an offer.  Additionally, UAs MUST treat
>the exchange of a 2xx response to an INVITE request and its
>corresponding ACK request as an offer/answer exchange."

IMO in this case, if a 2xx to a reINVITE is not an offer nor
an answer, a UAS may accept an UPDATE.
At least from an offer-answer perspective, the above additional
rule is unnecessary.

Regards,
Shinji

>"Similarly, if an UPDATE is received that contains an offer, and
>the UAS has received an offer (in an UPDATE, PRACK, or INVITE)
>to which it has not yet generated an answer, the UAS MUST reject
>the UPDATE with a 500 response, and MUST include a Retry-After
>header field with a randomly chosen value between 0 and 10 seconds."

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From john.elwell@siemens-enterprise.com  Mon May 24 02:41:44 2010
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Brett Tate <brett@broadsoft.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Mon, 24 May 2010 11:41:33 +0200
Thread-Topic: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without	SDP
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without	SDP
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=20

> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Brett Tate
> Sent: 21 May 2010 18:34
> To: sipcore@ietf.org
> Subject: [sipcore] draft-ietf-sipcore-reinvite-04: glare and=20
> UPDATE without SDP
>=20
> Howdy,
>=20
> Draft-ietf-sipcore-reinvite-04 sections 3.5 and 4.5 indicate=20
> that an UPDATE without SDP should be treated (from a glare=20
> perspective) as though it contained an offer SDP.  However I=20
> currently find the definition of "glare" and the expected=20
> behavior to be ambiguous.
>=20
> What is the precise meaning of glare?  I ask mainly because=20
> some vendors send a 491 when an RFC indicates to send a 500.
>=20
> If INVITE contains offer SDP and 18x was not reliable, how=20
> SHOULD the called party respond if UPDATE without SDP is=20
> received from caller: 491, 500, or process as though no glare?
>=20
> If INVITE contains offer SDP and 18x was not reliable, how=20
> SHOULD the caller respond if UPDATE without SDP is received=20
> from called party: 491, 500, or process as though no glare?
[JRE] I believe these two situations are both valid and should not result i=
n a 4xx/5xx/6xx response unless something else is wrong with the request.

John


>=20
> If caller receives INVITE 2xx without offer/answer, how=20
> SHOULD or MUST the caller respond if UPDATE without SDP is=20
> received from called party prior to sending ACK (if not=20
> willing to queue or drop UPDATE): 491 or 500?  Is it a SHOULD or MUST?
>=20
>=20
> Thanks,
> Brett
>=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =

From brett@broadsoft.com  Mon May 24 04:56:05 2010
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From: Brett Tate <brett@broadsoft.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Mon, 24 May 2010 04:54:51 -0700
Thread-Topic: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without	SDP
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> > Draft-ietf-sipcore-reinvite-04 sections 3.5 and 4.5 indicate
> > that an UPDATE without SDP should be treated (from a glare
> > perspective) as though it contained an offer SDP.  However I
> > currently find the definition of "glare" and the expected
> > behavior to be ambiguous.
> >
> > What is the precise meaning of glare?  I ask mainly because
> > some vendors send a 491 when an RFC indicates to send a 500.
> >
> > If INVITE contains offer SDP and 18x was not reliable, how
> > SHOULD the called party respond if UPDATE without SDP is
> > received from caller: 491, 500, or process as though no glare?
> >
> > If INVITE contains offer SDP and 18x was not reliable, how
> > SHOULD the caller respond if UPDATE without SDP is received
> > from called party: 491, 500, or process as though no glare?
>
> [JRE] I believe these two situations are both valid and should not
> result in a 4xx/5xx/6xx response unless something else is wrong with
> the request.

That was true before draft-ietf-sipcore-reinvite.  However it looks like th=
e following snippets make it no longer true.

Draft-ietf-sipcore-reinvite section 3.5:

"An UPDATE request without an offer can change dialog parameters.  So
 can a non-2xx final response to a re-INVITE request or a 2xx response
 to an INVITE request (re-INVITE or initial INVITE).  However, there
 are no rules to avoid a collision between an offerless UPDATE request
 and a final response to an INVITE request.  The rules in Section 4.5,
 which are given in the context of target refreshes, cover these types
 of collisions as well.  Therefore, there is no need to specify
 further rules here."

Draft-ietf-sipcore-reinvite section 4.5:

"When checking for glare situations, UAs MUST treat every UPDATE
 request as if it contained an offer."

RFC3311 section 5.2:

"If an UPDATE is received that contains an offer, and the UAS has
 generated an offer (in an UPDATE, PRACK or INVITE) to which it has
 not yet received an answer, the UAS MUST reject the UPDATE with a 491
 response.  Similarly, if an UPDATE is received that contains an
 offer, and the UAS has received an offer (in an UPDATE, PRACK, or
 INVITE) to which it has not yet generated an answer, the UAS MUST
 reject the UPDATE with a 500 response, and MUST include a Retry-After
 header field with a randomly chosen value between 0 and 10 seconds."


From brett@broadsoft.com  Mon May 24 05:07:39 2010
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From: Brett Tate <brett@broadsoft.com>
To: OKUMURA Shinji <shinji.okumura@softfront.jp>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Mon, 24 May 2010 05:06:28 -0700
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> >If I understand draft-ietf-sipcore-reinvite and RFC3311 correctly,
> >the following snippets apply.
> >
> >"When checking for glare situations, UAs MUST treat every UPDATE
> >request as if it contained an offer.  Additionally, UAs MUST treat
> >the exchange of a 2xx response to an INVITE request and its
> >corresponding ACK request as an offer/answer exchange."
>=20
> IMO in this case, if a 2xx to a reINVITE is not an offer nor
> an answer, a UAS may accept an UPDATE.
> At least from an offer-answer perspective, the above additional
> rule is unnecessary.

Draft-ietf-sipcore-reinvite has changed the offer/answer rules by forcing o=
ffer/answer rules upon requests/responses which don't actually contain an o=
ffer or answer.

And for completeness, draft-ietf-sipcore-reinvite (such as the above snippe=
t) has impacts on both INVITE and re-INVITE.


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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE withoutSDP
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From john.elwell@siemens-enterprise.com  Mon May 24 05:23:16 2010
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Brett Tate <brett@broadsoft.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Mon, 24 May 2010 14:22:44 +0200
Thread-Topic: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without	SDP
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without	SDP
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Brett,

Thanks for the clarification. So the changes do not prohibit the use of UPD=
ATE after only a non-reliable provisional response, provided the UPDATE req=
uest does not contain an offer - correct? If it does contain an offer it is=
 subject to the glare rules.

John=20

> -----Original Message-----
> From: Brett Tate [mailto:brett@broadsoft.com]=20
> Sent: 24 May 2010 12:55
> To: Elwell, John; sipcore@ietf.org
> Subject: RE: [sipcore] draft-ietf-sipcore-reinvite-04: glare=20
> and UPDATE without SDP
>=20
> > > Draft-ietf-sipcore-reinvite-04 sections 3.5 and 4.5 indicate
> > > that an UPDATE without SDP should be treated (from a glare
> > > perspective) as though it contained an offer SDP.  However I
> > > currently find the definition of "glare" and the expected
> > > behavior to be ambiguous.
> > >
> > > What is the precise meaning of glare?  I ask mainly because
> > > some vendors send a 491 when an RFC indicates to send a 500.
> > >
> > > If INVITE contains offer SDP and 18x was not reliable, how
> > > SHOULD the called party respond if UPDATE without SDP is
> > > received from caller: 491, 500, or process as though no glare?
> > >
> > > If INVITE contains offer SDP and 18x was not reliable, how
> > > SHOULD the caller respond if UPDATE without SDP is received
> > > from called party: 491, 500, or process as though no glare?
> >
> > [JRE] I believe these two situations are both valid and should not
> > result in a 4xx/5xx/6xx response unless something else is wrong with
> > the request.
>=20
> That was true before draft-ietf-sipcore-reinvite.  However it=20
> looks like the following snippets make it no longer true.
>=20
> Draft-ietf-sipcore-reinvite section 3.5:
>=20
> "An UPDATE request without an offer can change dialog parameters.  So
>  can a non-2xx final response to a re-INVITE request or a 2xx response
>  to an INVITE request (re-INVITE or initial INVITE).  However, there
>  are no rules to avoid a collision between an offerless UPDATE request
>  and a final response to an INVITE request.  The rules in Section 4.5,
>  which are given in the context of target refreshes, cover these types
>  of collisions as well.  Therefore, there is no need to specify
>  further rules here."
>=20
> Draft-ietf-sipcore-reinvite section 4.5:
>=20
> "When checking for glare situations, UAs MUST treat every UPDATE
>  request as if it contained an offer."
>=20
> RFC3311 section 5.2:
>=20
> "If an UPDATE is received that contains an offer, and the UAS has
>  generated an offer (in an UPDATE, PRACK or INVITE) to which it has
>  not yet received an answer, the UAS MUST reject the UPDATE with a 491
>  response.  Similarly, if an UPDATE is received that contains an
>  offer, and the UAS has received an offer (in an UPDATE, PRACK, or
>  INVITE) to which it has not yet generated an answer, the UAS MUST
>  reject the UPDATE with a 500 response, and MUST include a Retry-After
>  header field with a randomly chosen value between 0 and 10 seconds."
>=20
> =

From brett@broadsoft.com  Mon May 24 05:24:24 2010
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From: Brett Tate <brett@broadsoft.com>
To: "gao.yang2@zte.com.cn" <gao.yang2@zte.com.cn>
Date: Mon, 24 May 2010 05:23:09 -0700
Thread-Topic: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE withoutSDP
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> While UAS(as this is about Re-INVITE, UAS could be more proper here)=20

Where does it say draft-ietf-sipcore-reinvite sections 3.5 and 4.5 are limi=
ted to re-INVITE?  Notice section 3.5 explicitly indicates "re-INVITE or in=
itial INVITE".

Section 3.5 indicates:
"An UPDATE request without an offer can change dialog parameters.  So
 can a non-2xx final response to a re-INVITE request or a 2xx response
 to an INVITE request (re-INVITE or initial INVITE).  However, there
 are no rules to avoid a collision between an offerless UPDATE request
 and a final response to an INVITE request.  The rules in Section 4.5,
 which are given in the context of target refreshes, cover these types
 of collisions as well.  Therefore, there is no need to specify
 further rules here."

<snip>


> This is no a glare situation. UAC can handle the UPDATE OK.=20

Per draft-ietf-sipcore-reinvite sections 3.5 and 4.5 it is a 500 response s=
ituation.  The "offer" was within INVITE 2xx even though it contained no SD=
P.  Another "offer" was within the UPDATE even though it contained no SDP. =
 Thus per RFC3311, a 500 with Retry-After MUST be returned.



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From brett@broadsoft.com  Mon May 24 05:48:14 2010
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Thread-Topic: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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> Such rules in the text's 3.5 and 4.5, are used for=20
> glare situations, which have ambiguity.=20
>
> So, it is not want to override current 3261 and 3311's=20
> semantics. It is UPDATE and clarification for the *
> dark corner*, which is ambiguous case at all.=20

Draft-ietf-sipcore-reinvite section 3.5 and 4.5 appear to have changed the =
offer/answer rules by mandating that the rules apply when some requests and=
 responses don't actually contain an offer or answer.

Hopefully the authors will clarify the intended behavior of sections 3.5 an=
d 4.5 so that the text can be adjusted if needed.


From shinji.okumura@softfront.jp  Mon May 24 05:49:21 2010
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From: OKUMURA Shinji <shinji.okumura@softfront.jp>
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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Brett,

>> >If I understand draft-ietf-sipcore-reinvite and RFC3311 correctly,
>> >the following snippets apply.
>> >
>> >"When checking for glare situations, UAs MUST treat every UPDATE
>> >request as if it contained an offer.  Additionally, UAs MUST treat
>> >the exchange of a 2xx response to an INVITE request and its
>> >corresponding ACK request as an offer/answer exchange."
>> 
>> IMO in this case, if a 2xx to a reINVITE is not an offer nor
>> an answer, a UAS may accept an UPDATE.
>> At least from an offer-answer perspective, the above additional
>> rule is unnecessary.
>
>Draft-ietf-sipcore-reinvite has changed the offer/answer
>rules by forcing offer/answer rules upon requests/responses
>which don't actually contain an offer or answer.

I think I understand it probably.
What I make an issue is the validity of it.
I doubt whether the aproch is appropriate.

Regards,
Shinji

>And for completeness, draft-ietf-sipcore-reinvite (such as
>the above snippet) has impacts on both INVITE and re-INVITE.

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> Draft-ietf-sipcore-reinvite section 3.5 and 4.5 appear to have 
> changed the offer/answer rules by mandating that the rules apply 
> when some requests and responses don't actually contain an offer or 
answer.
> 
> Hopefully the authors will clarify the intended behavior of sections
> 3.5 and 4.5 so that the text can be adjusted if needed.

We are trying to fix ambiguous case in 3261 and 3311, not want to override 
well-balanced things.

So, if you think there's rules have impact on *once* well-balanced use 
cases, please show the use cases.

And if you have better way for the ambigunous cases, please show it too.

Thanks,

Gao


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<br><tt><font size=2><br>
&gt; Draft-ietf-sipcore-reinvite section 3.5 and 4.5 appear to have <br>
&gt; changed the offer/answer rules by mandating that the rules apply <br>
&gt; when some requests and responses don't actually contain an offer or
answer.<br>
&gt; <br>
&gt; Hopefully the authors will clarify the intended behavior of sections<br>
&gt; 3.5 and 4.5 so that the text can be adjusted if needed.<br>
</font></tt>
<br><font size=2 face="sans-serif">We are trying to fix ambiguous case
in 3261 and 3311, not want to override well-balanced things.</font>
<br>
<br><tt><font size=2>So, if you think there's rules have impact on *once*
</font></tt><font size=2 face="sans-serif">well-balanced</font><tt><font size=2>
use cases, please show the use cases.</font></tt>
<br>
<br><tt><font size=2>And if you have better way for the ambigunous cases,
please show it too.</font></tt>
<br>
<br><tt><font size=2>Thanks,</font></tt>
<br>
<br><tt><font size=2>Gao</font></tt>
<br><br><pre>
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From brett@broadsoft.com  Mon May 24 06:03:54 2010
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> Thanks for the clarification.=20
> So the changes do not prohibit the use of
> UPDATE after only a non-reliable provisional response, provided the
> UPDATE request does not contain an offer - correct? If it does contain
> an offer it is subject to the glare rules.

Concerning glare, my understanding is that the draft mandates treating UPDA=
TE without offer as though it contained an offer.

I'm still trying to understand how section 3.5 and 4.5 are to be interprete=
d.  Because the draft uses offer/answer rules for the "fix", it prevents us=
ing UPDATE without SDP on an early dialog until allowed per offer/answer ru=
les.  Thus sending 18x using 100rel with answer SDP is now important to avo=
id blocking ability to send UPDATE for non SDP reasons.


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> Concerning glare, my understanding is that the draft mandates 
> treating UPDATE without offer as though it contained an offer.
> 
> I'm still trying to understand how section 3.5 and 4.5 are to be 
> interpreted.  Because the draft uses offer/answer rules for the 
> "fix", it prevents using UPDATE without SDP on an early dialog until
> allowed per offer/answer rules.

Not exactly here. The text never forbit usage of UPDATE without offer by 
O/A rules.

People can use it as before, and just refer the rules here while it is a 
really glare case.


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<br><tt><font size=2><br>
&gt; Concerning glare, my understanding is that the draft mandates <br>
&gt; treating UPDATE without offer as though it contained an offer.<br>
&gt; <br>
&gt; I'm still trying to understand how section 3.5 and 4.5 are to be <br>
&gt; interpreted. &nbsp;Because the draft uses offer/answer rules for the
<br>
&gt; &quot;fix&quot;, it prevents using UPDATE without SDP on an early
dialog until<br>
&gt; allowed per offer/answer rules.</font></tt>
<br>
<br><tt><font size=2>Not exactly here. The text never forbit usage of UPDATE
without offer by O/A rules.</font></tt>
<br>
<br><tt><font size=2>People can use it as before, and just refer the rules
here while it is a really glare case.<br>
</font></tt><br><pre>
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From: Brett Tate <brett@broadsoft.com>
To: OKUMURA Shinji <shinji.okumura@softfront.jp>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Mon, 24 May 2010 06:15:42 -0700
Thread-Topic: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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> What I make an issue is the validity of it.
> I doubt whether the aproch is appropriate.

The draft has already passed WGLC.  The current status is "Publication Requ=
ested".

Thus I'm not sure how willing the authors and working group would be to alt=
er the solution.  However I'm not sure how many working group members notic=
ed the impacts of section 3.5 and 4.5.

It definitely encourages sending answer SDP within reliable 18x.  Not doing=
 so would block ability to use UPDATE on the early dialog.


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From brett@broadsoft.com  Mon May 24 06:26:39 2010
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From: Brett Tate <brett@broadsoft.com>
To: "gao.yang2@zte.com.cn" <gao.yang2@zte.com.cn>
Date: Mon, 24 May 2010 06:25:28 -0700
Thread-Topic: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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> > Concerning glare, my understanding is that the draft mandates=20
> > treating UPDATE without offer as though it contained an offer.
> >=20
> > I'm still trying to understand how section 3.5 and 4.5 are to be=20
> > interpreted. =A0Because the draft uses offer/answer rules for the=20
> > "fix", it prevents using UPDATE without SDP on an early dialog until
> > allowed per offer/answer rules.=20
>
> Not exactly here. The text never forbit usage of UPDATE without offer by =
O/A rules.=20

I agree that it doesn't forbid sending it.  It allows it and mandates the U=
AS return 491 or 500 (depending upon direction).


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From brett@broadsoft.com  Mon May 24 06:49:01 2010
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From: Brett Tate <brett@broadsoft.com>
To: "gao.yang2@zte.com.cn" <gao.yang2@zte.com.cn>
Date: Mon, 24 May 2010 06:47:48 -0700
Thread-Topic: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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> Sorry, I can't catch the meaning of "when you factor=20
> in RFC3311". Could you show me the meaning?

Look at RFC3311 (especially section 5.2) concerning how to behave when UPDA=
TE offer is received while there is an outstanding offer.

" If an UPDATE is received that contains an offer, and the UAS has
 generated an offer (in an UPDATE, PRACK or INVITE) to which it has
 not yet received an answer, the UAS MUST reject the UPDATE with a 491
 response.  Similarly, if an UPDATE is received that contains an
 offer, and the UAS has received an offer (in an UPDATE, PRACK, or
 INVITE) to which it has not yet generated an answer, the UAS MUST
 reject the UPDATE with a 500 response, and MUST include a Retry-After
 header field with a randomly chosen value between 0 and 10 seconds."



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From brett@broadsoft.com  Mon May 24 07:19:21 2010
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From: Brett Tate <brett@broadsoft.com>
To: "gao.yang2@zte.com.cn" <gao.yang2@zte.com.cn>
Date: Mon, 24 May 2010 07:18:10 -0700
Thread-Topic: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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> > Look at RFC3311 (especially section 5.2) concerning how to behave=20
> > when UPDATE offer is received while there is an outstanding offer.
>=20
> I know this part of 3311. I don't know the meaning of why=20
> you highlight this part.=20
>
> Do you think text of our draft has conflict with this part of 3311 here?=
=20
>
> I don't think so, as our text only for ambiguous cases.=20

I'm indicating that the RFC3311 mandates the returning 491 or 500 (dependin=
g upon direction).

You are indicating that the RFC3311 text doesn't apply even though draft-ie=
tf-sipcore-reinvite-04 sections 3.5 and 4.5 indicate they would apply.  RFC=
3311 section 5.2 includes glare checking as part of normal UPDATE processin=
g.

The drafts normative text causing the issue: "When checking for glare situa=
tions, UAs MUST treat every UPDATE request as if it contained an offer."

You keep interpreting/indicating that the normative text does not apply to =
all UPDATEs.  Please provide the normative text which indicates your interp=
retation.

Thanks,
Brett


From brett@broadsoft.com  Mon May 24 07:28:48 2010
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From: Brett Tate <brett@broadsoft.com>
To: "sipcore@ietf.org" <sipcore@ietf.org>, "Gonzalo.Camarillo@ericsson.com" <Gonzalo.Camarillo@ericsson.com>, "Christer.Holmberg@ericsson.com" <Christer.Holmberg@ericsson.com>, "gao.yang2@zte.com.cn" <gao.yang2@zte.com.cn>
Date: Mon, 24 May 2010 07:27:37 -0700
Thread-Topic: draft-ietf-sipcore-reinvite-04: updates rfc3311
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Subject: [sipcore] draft-ietf-sipcore-reinvite-04: updates rfc3311
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Doesn't draft-ietf-sipcore-reinvite-04 update RFC3311?

For instance section 4.5 indicates the following: "When checking for glare =
situations, UAs MUST treat every UPDATE request as if it contained an offer=
."

If the answer is yes, does it need to be listed on first page next to 3261?

Thanks,
Brett


From gao.yang2@zte.com.cn  Mon May 24 07:37:48 2010
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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Rules for "When checking for glare situations" in the draft text means the 
UAs's reasoning and handling while the UAs are in a glare situations, 
which is ambiguous.

It does not means UAs reasoning and handling for every use case to figure 
out whether this case is a glare case.

This is my understanding. And I'd like other co-authors and important 
reviewer to comment on this point. 

> > > Look at RFC3311 (especially section 5.2) concerning how to behave 
> > > when UPDATE offer is received while there is an outstanding offer.
> > 
> > I know this part of 3311. I don't know the meaning of why 
> > you highlight this part. 
> >
> > Do you think text of our draft has conflict with this part of 3311 
here? 
> >
> > I don't think so, as our text only for ambiguous cases. 
> 
> I'm indicating that the RFC3311 mandates the returning 491 or 500 
> (depending upon direction).
> 
> You are indicating that the RFC3311 text doesn't apply even though 
> draft-ietf-sipcore-reinvite-04 sections 3.5 and 4.5 indicate they 
> would apply.  RFC3311 section 5.2 includes glare checking as part of
> normal UPDATE processing.
> 
> The drafts normative text causing the issue: "When checking for 
> glare situations, UAs MUST treat every UPDATE request as if it 
> contained an offer."
> 
> You keep interpreting/indicating that the normative text does not 
> apply to all UPDATEs.  Please provide the normative text which 
> indicates your interpretation.
> 
> Thanks,
> Brett
> 
> 


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<br><tt><font size=2>Rules for &quot;</font></tt><font size=2 face="Courier New">When
checking for glare situations</font><tt><font size=2>&quot; in the draft
text means the UAs's reasoning and handling while the UAs are in a </font></tt><font size=2 face="Courier New">glare
situations</font><tt><font size=2>, which is ambiguous.</font></tt>
<br>
<br><tt><font size=2>It does not means UAs reasoning and handling for every
use case to figure out whether this case is a glare case.</font></tt>
<br>
<br><tt><font size=2>This is my understanding. And I'd like other co-authors
and important reviewer to comment on this point. </font></tt>
<br><tt><font size=2><br>
&gt; &gt; &gt; Look at RFC3311 (especially section 5.2) concerning how
to behave <br>
&gt; &gt; &gt; when UPDATE offer is received while there is an outstanding
offer.<br>
&gt; &gt; <br>
&gt; &gt; I know this part of 3311. I don't know the meaning of why <br>
&gt; &gt; you highlight this part. <br>
&gt; &gt;<br>
&gt; &gt; Do you think text of our draft has conflict with this part of
3311 here? <br>
&gt; &gt;<br>
&gt; &gt; I don't think so, as our text only for ambiguous cases. <br>
&gt; <br>
&gt; I'm indicating that the RFC3311 mandates the returning 491 or 500
<br>
&gt; (depending upon direction).<br>
&gt; <br>
&gt; You are indicating that the RFC3311 text doesn't apply even though
<br>
&gt; draft-ietf-sipcore-reinvite-04 sections 3.5 and 4.5 indicate they
<br>
&gt; would apply. &nbsp;RFC3311 section 5.2 includes glare checking as
part of<br>
&gt; normal UPDATE processing.<br>
&gt; <br>
&gt; The drafts normative text causing the issue: &quot;When checking for
<br>
&gt; glare situations, UAs MUST treat every UPDATE request as if it <br>
&gt; contained an offer.&quot;<br>
&gt; <br>
&gt; You keep interpreting/indicating that the normative text does not
<br>
&gt; apply to all UPDATEs. &nbsp;Please provide the normative text which
<br>
&gt; indicates your interpretation.<br>
&gt; <br>
&gt; Thanks,<br>
&gt; Brett<br>
&gt; <br>
&gt; <br>
</font></tt><br><pre>
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</pre>
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From brett@broadsoft.com  Mon May 24 08:03:28 2010
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Subject: [sipcore] draft-ietf-sipcore-reinvite-04: impacts upon REFER and NOTIFY
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Since in-dialog REFER and NOTIFY are target refresh requests, what are the =
impacts of section 4.5 "Race Conditions and Target Refreshes" upon REFER an=
d NOTIFY?

It doesn't look like there are any normative impacts.  However I wasn't sur=
e if this was intentional.

Thanks,
Brett



From john.elwell@siemens-enterprise.com  Mon May 24 08:12:46 2010
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To: Brett Tate <brett@broadsoft.com>, "gao.yang2@zte.com.cn" <gao.yang2@zte.com.cn>
Date: Mon, 24 May 2010 17:12:35 +0200
Thread-Topic: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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If an UPDATE request without SDP offer cannot be accepted when only an unre=
liable provisional response has occurred, I believe this is a normative cha=
nge to RFC 3311.

John

> -----Original Message-----
> From: Brett Tate [mailto:brett@broadsoft.com]=20
> Sent: 24 May 2010 14:25
> To: gao.yang2@zte.com.cn
> Cc: Elwell, John; sipcore@ietf.org
> Subject: RE: [sipcore] draft-ietf-sipcore-reinvite-04: glare=20
> and UPDATE without SDP
>=20
> > > Concerning glare, my understanding is that the draft mandates=20
> > > treating UPDATE without offer as though it contained an offer.
> > >=20
> > > I'm still trying to understand how section 3.5 and 4.5 are to be=20
> > > interpreted. =A0Because the draft uses offer/answer rules for the=20
> > > "fix", it prevents using UPDATE without SDP on an early=20
> dialog until
> > > allowed per offer/answer rules.=20
> >
> > Not exactly here. The text never forbit usage of UPDATE=20
> without offer by O/A rules.=20
>=20
> I agree that it doesn't forbid sending it.  It allows it and=20
> mandates the UAS return 491 or 500 (depending upon direction).
>=20
> =

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From christer.holmberg@ericsson.com  Mon May 24 09:56:11 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: SIPCORE <sipcore@ietf.org>
Date: Mon, 24 May 2010 18:55:57 +0200
Thread-Topic: Draft new version: draft-ietf-sipcore-keep-03
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Subject: [sipcore] Draft new version: draft-ietf-sipcore-keep-03
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Hi,

I've submitted a new version of the keep draft.

Based on feedback and comments, I've moved back to the original Via paramet=
er based solution.

I've done a number of editorial changes in order to make the draft more cle=
ar.

The currently only OPEN ISSUE is whether we should use the Flow-Timer heade=
r field, defined in Outbound, to indicate the keep-alive frequency, or whet=
her the frequency should be added directly to the Via parameter as a parame=
ter value (instead of adding a "yes" value).

Using Flow-Timer is alligned with Outbound, while using a keep parameter va=
lue is more "compact".

Flow-Timer example:   'Flow-Timer: 30'

Via keep parameter example: Via:11.22.33.44;keep=3D30

The keep-draft allows usage of the Flow-Timer header field also in initial =
requests for dialogs, and mid-dialog target refresh requests, but I did not=
 find any text in the Outbound spec which would restrict the usage of the h=
eader field to REGISTER requests only.

Once we've agreed on how to move forward with the keep-alive frequency issu=
e I believe the draft is ready for WGLC.

Regards,

Christer

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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           : Indication of support for keep-alive
	Author(s)       : C. Holmberg
	Filename        : draft-ietf-sipcore-keep-03.txt
	Pages           : 15
	Date            : 2010-05-24

This specification defines a new Session Initiation Protocol (SIP)
Via header field parameter, "keep", which allows adjacent SIP
entities to explicitly negotiate usage of the Network Address
Translation (NAT) keep-alive mechanisms defined in SIP Outbound, in
cases where SIP Outbound is not supported, cannot be applied, or
where usage of keep-alives are not implicitly negotiated as part of
the SIP Outbound negotiation.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sipcore-keep-03.txt

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ftp://ftp.ietf.org/internet-drafts/

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From shinji.okumura@softfront.jp  Mon May 24 20:27:25 2010
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From: OKUMURA Shinji <shinji.okumura@softfront.jp>
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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Brett,

Sorry for my lack of experience for publication of ID.
Was there an announcement about WGLC in this ML?

In any case, I do not deny a basic concept.
I agree that UAC generate new offer to synchronize.
But I doubt that UAs treat the final response to INVITE
and ACK as an offer/answer exchange.

Perhaps I think that my opinion is similar to yours.

Regards,
Shinji

Brett Tate <brett@broadsoft.com>
Mon, 24 May 2010 06:15:42 -0700
>> What I make an issue is the validity of it.
>> I doubt whether the approach is appropriate.
>
>The draft has already passed WGLC.  The current status is
>"Publication Requested".
>
>Thus I'm not sure how willing the authors and working group
>would be to alter the solution.  However I'm not sure how many
>working group members noticed the impacts of section 3.5 and 4.5.
>
>It definitely encourages sending answer SDP within reliable 18x.
>Not doing so would block ability to use UPDATE on the early dialog.

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From: OKUMURA Shinji <shinji.okumura@softfront.jp>
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATEwithout SDP
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Gao,

gao.yang2@zte.com.cn
Mon, 24 May 2010 21:30:21 +0800
>
>> > > Concerning glare, my understanding is that the draft mandates 
>> > > treating UPDATE without offer as though it contained an offer.
>> > > 
>> > > I'm still trying to understand how section 3.5 and 4.5 are to be 
>> > > interpreted.  Because the draft uses offer/answer rules for the 
>> > > "fix", it prevents using UPDATE without SDP on an early dialog until
>> > > allowed per offer/answer rules. 
>> >
>> > Not exactly here. The text never forbit usage of UPDATE without 
>> offer by O/A rules. 
>> 
>> I agree that it doesn't forbid sending it.  It allows it and 
>> mandates the UAS return 491 or 500 (depending upon direction).
>
>It mandates the UAS send 4xx/5xx to UPDATE only when the UAS
>send 4xx/5xx to the Re-INVITE and the UAS get the UPDATE before
>ACK of 4xx/5xx(Re-INVITE).

As you know, ACK to non-2xx is hop-by-hop.
There is little benefit to reject such an UPDATE.

    A                               B
    |                               |
  s0|re-INV (offer1)                |s0
    |------------------------------>|
    |                   18x(answer1)|
    |<------------------------------|
  s1|                        non-2xx|s1
    |<------------------------------|<-+
  s0|ACK                            |s0|
    |----------->                   |  |
    |UPD(offer0)                    |  |
    |==============================>|* | reject UPDATE
    |                       491(UPD)|  |
    |<==============================|  |
    |                    ACK        |  |
    |                    ---------->|<-+
    |                               |
                Figure 2

    A                               B
    |                               |
  s0|re-INV (offer1)                |s0
    |------------------------------>|
    |                   18x(answer1)|
    |<------------------------------|
  s1|                        non-2xx|s1
    |                 /-------------|<-+
    |                /   ACK        |s0| reject UPDATE
    |               /    ---------->|<-+
    |UPD(offer2)   /                |
    |=============/================>|
    |            /          200(UPD)|
    |<==========/===================|
  s2|          /                    |s2
    |<--------/                     |
  s0|ACK                            |
    |----------->                   |
    |                               |
                Figure 3

I've also been pointed out by Ian Elz.

Regards,
Shinji

>If UAS send 2xx to Re-INVITE, the UAS would not send 4xx/5xx
>to the UPDATE.
>
>And if the UAS get the ACK(Re-INVITE's 4xx/5xx) firstly, the
>UAS also would not mandate to the rules.

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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATEwithout SDP
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Hi,


> >> I agree that it doesn't forbid sending it.  It allows it and 
> >> mandates the UAS return 491 or 500 (depending upon direction).
> >
> >It mandates the UAS send 4xx/5xx to UPDATE only when the UAS
> >send 4xx/5xx to the Re-INVITE and the UAS get the UPDATE before
> >ACK of 4xx/5xx(Re-INVITE).
> 
> As you know, ACK to non-2xx is hop-by-hop.
> There is little benefit to reject such an UPDATE.

UAS send 491 for the UPDATE(in the flow) to *tell* UAC that in it's(UAS) 
view, there's a glare case.

When UAS get the ACK then UPDATE, it still has no precise criterion for 
whether the UAC gets non-2xx(Re-INVITE) first then send the UPDATE or not. 
Keeping rejecting the UPDATE is not a reasonable way here. So, the draft 
suggest the way, as UAS can send 2xx to the UPDATE.

When UAC get UAS's 4xx of UPDATE, the session state is clear for it. And 
UAC would only send a synchronizing Re-INVITE/UPDATE when it is a glare 
case in the UAC's view.

IMO, this way is better than the way we might once use: allowing B send 
2xx to UPDATE, and both A and B *see* glare, and all send Re-INVITE/UPDATE 
to refresh session state by each's view.

Thanks,

Gao

> 
>     A                               B
>     |                               |
>   s0|re-INV (offer1)                |s0
>     |------------------------------>|
>     |                   18x(answer1)|
>     |<------------------------------|
>   s1|                        non-2xx|s1
>     |<------------------------------|<-+
>   s0|ACK                            |s0|
>     |----------->                   |  |
>     |UPD(offer0)                    |  |
>     |==============================>|* | reject UPDATE
>     |                       491(UPD)|  |
>     |<==============================|  |
>     |                    ACK        |  |
>     |                    ---------->|<-+
>     |                               |
>                 Figure 2
> 
>     A                               B
>     |                               |
>   s0|re-INV (offer1)                |s0
>     |------------------------------>|
>     |                   18x(answer1)|
>     |<------------------------------|
>   s1|                        non-2xx|s1
>     |                 /-------------|<-+
>     |                /   ACK        |s0| reject UPDATE
>     |               /    ---------->|<-+
>     |UPD(offer2)   /                |
>     |=============/================>|
>     |            /          200(UPD)|
>     |<==========/===================|
>   s2|          /                    |s2
>     |<--------/                     |
>   s0|ACK                            |
>     |----------->                   |
>     |                               |
>                 Figure 3
> 
> I've also been pointed out by Ian Elz.
> 
> Regards,
> Shinji
> 
> >If UAS send 2xx to Re-INVITE, the UAS would not send 4xx/5xx
> >to the UPDATE.
> >
> >And if the UAS get the ACK(Re-INVITE's 4xx/5xx) firstly, the
> >UAS also would not mandate to the rules.
> 


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<br><font size=2 face="sans-serif">Hi,</font>
<br>
<br><tt><font size=2><br>
&gt; &gt;&gt; I agree that it doesn't forbid sending it. &nbsp;It allows
it and <br>
&gt; &gt;&gt; mandates the UAS return 491 or 500 (depending upon direction).<br>
&gt; &gt;<br>
&gt; &gt;It mandates the UAS send 4xx/5xx to UPDATE only when the UAS<br>
&gt; &gt;send 4xx/5xx to the Re-INVITE and the UAS get the UPDATE before<br>
&gt; &gt;ACK of 4xx/5xx(Re-INVITE).<br>
&gt; <br>
&gt; As you know, ACK to non-2xx is hop-by-hop.<br>
&gt; There is little benefit to reject such an UPDATE.</font></tt>
<br>
<br><tt><font size=2>UAS send 491 for the UPDATE(in the flow) to *tell*
UAC that in it's(UAS) view, there's a glare case.</font></tt>
<br>
<br><tt><font size=2>When UAS get the ACK then UPDATE, it still has no
precise criterion for whether the UAC gets non-2xx(Re-INVITE) first then
send the UPDATE or not. Keeping rejecting the UPDATE is not a reasonable
way here. So, the draft suggest the way, as UAS can send 2xx to the UPDATE.</font></tt>
<br>
<br><tt><font size=2>When UAC get UAS's 4xx of UPDATE, the session state
is clear for it. And UAC would only send a synchronizing Re-INVITE/UPDATE
when it is a glare case in the UAC's view.</font></tt>
<br>
<br><tt><font size=2>IMO, this way is better than the way we might once
use: allowing B send 2xx to UPDATE, and both A and B *see* glare, and all
send Re-INVITE/UPDATE to refresh session state by each's view.</font></tt>
<br>
<br><tt><font size=2>Thanks,</font></tt>
<br>
<br><tt><font size=2>Gao</font></tt>
<br><tt><font size=2><br>
&gt; <br>
&gt; &nbsp; &nbsp; A &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; B<br>
&gt; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |<br>
&gt; &nbsp; s0|re-INV (offer1) &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp;|s0<br>
&gt; &nbsp; &nbsp; |------------------------------&gt;|<br>
&gt; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; 18x(answer1)|<br>
&gt; &nbsp; &nbsp; |&lt;------------------------------|<br>
&gt; &nbsp; s1| &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp;non-2xx|s1<br>
&gt; &nbsp; &nbsp; |&lt;------------------------------|&lt;-+<br>
&gt; &nbsp; s0|ACK &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|s0|<br>
&gt; &nbsp; &nbsp; |-----------&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; | &nbsp;|<br>
&gt; &nbsp; &nbsp; |UPD(offer0) &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp;| &nbsp;|<br>
&gt; &nbsp; &nbsp; |==============================&gt;|* | reject UPDATE<br>
&gt; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; 491(UPD)| &nbsp;|<br>
&gt; &nbsp; &nbsp; |&lt;==============================| &nbsp;|<br>
&gt; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp;ACK &nbsp; &nbsp; &nbsp; &nbsp;| &nbsp;|<br>
&gt; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp;----------&gt;|&lt;-+<br>
&gt; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |<br>
&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Figure 2<br>
&gt; <br>
&gt; &nbsp; &nbsp; A &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; B<br>
&gt; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |<br>
&gt; &nbsp; s0|re-INV (offer1) &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp;|s0<br>
&gt; &nbsp; &nbsp; |------------------------------&gt;|<br>
&gt; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; 18x(answer1)|<br>
&gt; &nbsp; &nbsp; |&lt;------------------------------|<br>
&gt; &nbsp; s1| &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp;non-2xx|s1<br>
&gt; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
/-------------|&lt;-+<br>
&gt; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;/
&nbsp; ACK &nbsp; &nbsp; &nbsp; &nbsp;|s0| reject UPDATE<br>
&gt; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; /
&nbsp; &nbsp;----------&gt;|&lt;-+<br>
&gt; &nbsp; &nbsp; |UPD(offer2) &nbsp; / &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp;|<br>
&gt; &nbsp; &nbsp; |=============/================&gt;|<br>
&gt; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;/ &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp;200(UPD)|<br>
&gt; &nbsp; &nbsp; |&lt;==========/===================|<br>
&gt; &nbsp; s2| &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;/ &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|s2<br>
&gt; &nbsp; &nbsp; |&lt;--------/ &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; |<br>
&gt; &nbsp; s0|ACK &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|<br>
&gt; &nbsp; &nbsp; |-----------&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; |<br>
&gt; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |<br>
&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Figure 3<br>
&gt; <br>
&gt; I've also been pointed out by Ian Elz.<br>
&gt; <br>
&gt; Regards,<br>
&gt; Shinji<br>
&gt; <br>
&gt; &gt;If UAS send 2xx to Re-INVITE, the UAS would not send 4xx/5xx<br>
&gt; &gt;to the UPDATE.<br>
&gt; &gt;<br>
&gt; &gt;And if the UAS get the ACK(Re-INVITE's 4xx/5xx) firstly, the<br>
&gt; &gt;UAS also would not mandate to the rules.<br>
&gt; <br>
</font></tt><br><pre>
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X-Mailer: Lotus Notes Release 7.0.1 January 17, 2006
Message-ID: <OF77911A47.29F6ACC4-ON4825772E.0024D951-4825772E.00265D32@zte.com.cn>
From: gao.yang2@zte.com.cn
Date: Tue, 25 May 2010 14:56:58 +0800
X-MIMETrack: Serialize by Router on notes_smtp/zte_ltd(Release 6.5.4|March 27, 2005) at 2010-05-25 14:59:22, Serialize complete at 2010-05-25 14:59:22
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>, "Christer.Holmberg@ericsson.com" <Christer.Holmberg@ericsson.com>
Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: impacts upon REFER and NOTIFY
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--=_alternative 00265D314825772E_=--


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From: OKUMURA Shinji <shinji.okumura@softfront.jp>
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare andUPDATEwithout SDP
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Gao,

gao.yang2@zte.com.cn
Tue, 25 May 2010 14:15:10 +0800
>Hi,
>
>> >> I agree that it doesn't forbid sending it.  It allows it and 
>> >> mandates the UAS return 491 or 500 (depending upon direction).
>> >
>> >It mandates the UAS send 4xx/5xx to UPDATE only when the UAS
>> >send 4xx/5xx to the Re-INVITE and the UAS get the UPDATE before
>> >ACK of 4xx/5xx(Re-INVITE).
>> 
>> As you know, ACK to non-2xx is hop-by-hop.
>> There is little benefit to reject such an UPDATE.
>
>UAS send 491 for the UPDATE(in the flow) to *tell* UAC that in it's(UAS) 
>view, there's a glare case.

But this UPDATE is sent in order to synchronize the common view.
If UAS return 2xx, then the situation goes into nonglare.

Probably another UPDATE will be sent, but this UPDATE
transaction is supposed to be avoided.

At the very start, the UAS rejects a reINVITE does not care
about this text. Is it necessary to let such a UAS follow a
new rule?

>When UAS get the ACK then UPDATE, it still has no precise criterion for 
>whether the UAC gets non-2xx(Re-INVITE) first then send the UPDATE or not. 
>Keeping rejecting the UPDATE is not a reasonable way here. So, the draft 
>suggest the way, as UAS can send 2xx to the UPDATE.

It is a conventional handling and enough for UAS.

>When UAC get UAS's 4xx of UPDATE, the session state is clear for it. And 
>UAC would only send a synchronizing Re-INVITE/UPDATE when it is a glare 
>case in the UAC's view.

UAS' behavior does not affect UAC' behavior directly.
UAC will do the needful regardless of whether UAS rejects
a reINVITE.

>IMO, this way is better than the way we might once use: allowing B send 
>2xx to UPDATE, and both A and B *see* glare, and all send Re-INVITE/UPDATE 
>to refresh session state by each's view.

The new rule for UAS does not seem to improve the situation.
The new rule for UAC seems to be just enough to synchronize
the view.

Regards,
Shinji

>Thanks,
>
>Gao
>
>> 
>>     A                               B
>>     |                               |
>>   s0|re-INV (offer1)                |s0
>>     |------------------------------>|
>>     |                   18x(answer1)|
>>     |<------------------------------|
>>   s1|                        non-2xx|s1
>>     |<------------------------------|<-+
>>   s0|ACK                            |s0|
>>     |----------->                   |  |
>>     |UPD(offer0)                    |  |
>>     |==============================>|* | reject UPDATE
>>     |                       491(UPD)|  |
>>     |<==============================|  |
>>     |                    ACK        |  |
>>     |                    ---------->|<-+
>>     |                               |
>>                 Figure 2
>> 
>>     A                               B
>>     |                               |
>>   s0|re-INV (offer1)                |s0
>>     |------------------------------>|
>>     |                   18x(answer1)|
>>     |<------------------------------|
>>   s1|                        non-2xx|s1
>>     |                 /-------------|<-+
>>     |                /   ACK        |s0| reject UPDATE
>>     |               /    ---------->|<-+
>>     |UPD(offer2)   /                |
>>     |=============/================>|
>>     |            /          200(UPD)|
>>     |<==========/===================|
>>   s2|          /                    |s2
>>     |<--------/                     |
>>   s0|ACK                            |
>>     |----------->                   |
>>     |                               |
>>                 Figure 3
>> 
>> I've also been pointed out by Ian Elz.
>> 
>> Regards,
>> Shinji
>> 
>> >If UAS send 2xx to Re-INVITE, the UAS would not send 4xx/5xx
>> >to the UPDATE.
>> >
>> >And if the UAS get the ACK(Re-INVITE's 4xx/5xx) firstly, the
>> >UAS also would not mandate to the rules.

From john.elwell@siemens-enterprise.com  Tue May 25 01:34:02 2010
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, SIPCORE <sipcore@ietf.org>
Date: Tue, 25 May 2010 10:33:51 +0200
Thread-Topic: Draft new version: draft-ietf-sipcore-keep-03
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Christer,

Thanks, this addresses concerns I had with the previous version.

Digging deeper into the issue with the Flow-Timer header field, I view it t=
his way. The "keep" parameter is in the Via header field, which has one ent=
ry per hop over which the request passes. Therefore the "keep" parameter is=
 clearly associated with a given hop. However, the Flow-Timer header field =
does not have the concept of one entry per hop. Therefore the header field =
is assumed to apply only to the current hop, and has to be removed before f=
orwarding a response over a hop on which keep-alive has not been requested,=
 or needs to have its value modified before forwarding a response over a ho=
p on which keep-alive has been requested. It seems odd that the two pieces =
of information used by this feature should be handled in two completely dif=
ferent ways.

Another difference is that when a proxy receives a "keep" parameter in the =
topmost Via header field entry, it adds "yes" as the value of that paramete=
r when forwarding that Via header entry (which will no longer be the topmos=
t entry and therefore will not be inspected by the next entity). Therefore =
when the Via header field is returned in the response, it will already have=
 the "yes" value in place when it crosses the hop to which the entry applie=
s. Therefore the proxy does not need to take special action concerning the =
Via header field when forwarding the response. In contrast, a proxy has to =
take special action with the Flow-Timer header field when forwarding a resp=
onse, as described above. The way "keep" is handled seems to take account o=
f transaction-stateless proxies, whereas the way Flow-Timer is handle does =
not. I am not sure, however, whether this mechanism makes sense for use wit=
h transaction-stateless proxies anyway, so perhaps this latter point is moo=
t.

John


> -----Original Message-----
> From: sipcore-bounces@ietf.org=20
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Christer Holmberg
> Sent: 24 May 2010 17:56
> To: SIPCORE
> Subject: [sipcore] Draft new version: draft-ietf-sipcore-keep-03
>=20
> Hi,
>=20
> I've submitted a new version of the keep draft.
>=20
> Based on feedback and comments, I've moved back to the=20
> original Via parameter based solution.
>=20
> I've done a number of editorial changes in order to make the=20
> draft more clear.
>=20
> The currently only OPEN ISSUE is whether we should use the=20
> Flow-Timer header field, defined in Outbound, to indicate the=20
> keep-alive frequency, or whether the frequency should be=20
> added directly to the Via parameter as a parameter value=20
> (instead of adding a "yes" value).
>=20
> Using Flow-Timer is alligned with Outbound, while using a=20
> keep parameter value is more "compact".
>=20
> Flow-Timer example:   'Flow-Timer: 30'
>=20
> Via keep parameter example: Via:11.22.33.44;keep=3D30
>=20
> The keep-draft allows usage of the Flow-Timer header field=20
> also in initial requests for dialogs, and mid-dialog target=20
> refresh requests, but I did not find any text in the Outbound=20
> spec which would restrict the usage of the header field to=20
> REGISTER requests only.
>=20
> Once we've agreed on how to move forward with the keep-alive=20
> frequency issue I believe the draft is ready for WGLC.
>=20
> Regards,
>=20
> Christer
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =

From christer.holmberg@ericsson.com  Tue May 25 02:36:27 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, SIPCORE <sipcore@ietf.org>
Date: Tue, 25 May 2010 11:36:11 +0200
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Subject: Re: [sipcore] Draft new version: draft-ietf-sipcore-keep-03
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Hi John,

>Digging deeper into the issue with the Flow-Timer header field, I view it =
this way. The "keep" parameter is in the Via header field, which has one en=
try per hop over which the request passes. Therefore the "keep" parameter i=
s clearly associated with a given hop. However,=20
>the Flow-Timer header field does not have the concept of one entry per hop=
. Therefore the header field is assumed to apply only to the current hop, a=
nd has to be removed before forwarding a response over a hop on which keep-=
alive has not been requested, or needs to=20
>have its value modified before forwarding a response over a hop on which k=
eep-alive has been requested.=20

True.=20

Of course, in Outbound that is not a problem, since Flow-Timer is only used=
 on one hop (between the UA and its edge proxy).

>It seems odd that the two pieces of information used by this feature shoul=
d be handled in two completely different ways.
>
>Another difference is that when a proxy receives a "keep" parameter in the=
 topmost Via header field entry, it adds "yes" as the value of that paramet=
er when forwarding that Via header entry (which will no longer be the topmo=
st entry and therefore will not be inspected by=20
>the next entity). Therefore when the Via header field is returned in the r=
esponse, it will already have the "yes" value in place when it crosses the =
hop to which the entry applies. Therefore the proxy does not need to take s=
pecial action concerning the Via header field when=20
>forwarding the response. In contrast, a proxy has to take special action w=
ith the Flow-Timer header field when forwarding a response, as described ab=
ove. The way "keep" is handled seems to take account of transaction-statele=
ss proxies, whereas the way Flow-Timer is=20
>handle does not. I am not sure, however, whether this mechanism makes sens=
e for use with transaction-stateless proxies anyway, so perhaps this latter=
 point is moot.

I think that proxies do need to be stateful - at least if they intend to se=
nd keep-alives within a dialog (I doubt a proxy would negotiate sending of =
proxies within a registration, since proxies are normally registration un-a=
ware). Because, the proxy needs to know when the dialog is terminated, in o=
rder to stop sending the keep-alives.

Also, a proxy that receives keep-alives also needs to be stateful in case i=
t intends to perform some actions in case negotiated keep-alives aren't rec=
eived within a dialog. The draft (nor Outbound, afaik) defines what a proxy=
 does in those cases, because it is seen as an implementation issue.

As said earlier, the reason for using Flow-Timer in the draft is because Ou=
tbound already uses it, and the purpose of the keep-draft is to not change =
the Outbound keep-alive mechanism as such, but rather define a mechanism to=
 explicitly negotiate usage of it.

But, personally I have no strong preference for which solution we use. Howe=
ver, IF we choose to indicate the frequency as a keep parameter value, I th=
ink we should add some text to the draft explaining why Flow-Timer is not u=
sed.

Regards,

Christer



> -----Original Message-----
> From: sipcore-bounces@ietf.org
> [mailto:sipcore-bounces@ietf.org] On Behalf Of Christer Holmberg
> Sent: 24 May 2010 17:56
> To: SIPCORE
> Subject: [sipcore] Draft new version: draft-ietf-sipcore-keep-03
>
> Hi,
>
> I've submitted a new version of the keep draft.
>
> Based on feedback and comments, I've moved back to the
> original Via parameter based solution.
>
> I've done a number of editorial changes in order to make the
> draft more clear.
>
> The currently only OPEN ISSUE is whether we should use the
> Flow-Timer header field, defined in Outbound, to indicate the
> keep-alive frequency, or whether the frequency should be
> added directly to the Via parameter as a parameter value
> (instead of adding a "yes" value).
>
> Using Flow-Timer is alligned with Outbound, while using a
> keep parameter value is more "compact".
>
> Flow-Timer example:   'Flow-Timer: 30'
>
> Via keep parameter example: Via:11.22.33.44;keep=3D30
>
> The keep-draft allows usage of the Flow-Timer header field
> also in initial requests for dialogs, and mid-dialog target
> refresh requests, but I did not find any text in the Outbound
> spec which would restrict the usage of the header field to
> REGISTER requests only.
>
> Once we've agreed on how to move forward with the keep-alive
> frequency issue I believe the draft is ready for WGLC.
>
> Regards,
>
> Christer
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
>=

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From brett@broadsoft.com  Tue May 25 04:39:13 2010
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From: Brett Tate <brett@broadsoft.com>
To: OKUMURA Shinji <shinji.okumura@softfront.jp>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Tue, 25 May 2010 04:37:59 -0700
Thread-Topic: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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Yes; there was a WGLC announcement.

http://www.ietf.org/mail-archive/web/sipcore/current/msg01950.html=20

http://www.ietf.org/mail-archive/web/sipcore/current/msg02089.html=20


Even though I may not prefer the solution (as I understand it) mandated by =
sections 3.5 are 4.5, I'm mainly just trying to understand it.  Gao and I c=
urrently don't agree upon how sections 3.5 and 4.5 impact UPDATE without SD=
P on an early dialog.  As Gao has requested, I hope the other authors and P=
aul will reply.

http://www.ietf.org/mail-archive/web/sipcore/current/msg02750.html=20


> -----Original Message-----
> From: OKUMURA Shinji [mailto:shinji.okumura@softfront.jp]
> Sent: Monday, May 24, 2010 11:22 PM
> To: sipcore@ietf.org
> Cc: Brett Tate
> Subject: RE: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE
> without SDP
>=20
> Brett,
>=20
> Sorry for my lack of experience for publication of ID.
> Was there an announcement about WGLC in this ML?
>=20
> In any case, I do not deny a basic concept.
> I agree that UAC generate new offer to synchronize.
> But I doubt that UAs treat the final response to INVITE
> and ACK as an offer/answer exchange.
>=20
> Perhaps I think that my opinion is similar to yours.
>=20
> Regards,
> Shinji
>=20
> Brett Tate <brett@broadsoft.com>
> Mon, 24 May 2010 06:15:42 -0700
> >> What I make an issue is the validity of it.
> >> I doubt whether the approach is appropriate.
> >
> >The draft has already passed WGLC.  The current status is
> >"Publication Requested".
> >
> >Thus I'm not sure how willing the authors and working group
> >would be to alter the solution.  However I'm not sure how many
> >working group members noticed the impacts of section 3.5 and 4.5.
> >
> >It definitely encourages sending answer SDP within reliable 18x.
> >Not doing so would block ability to use UPDATE on the early dialog.

From shinji.okumura@softfront.jp  Tue May 25 07:36:45 2010
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From: OKUMURA Shinji <shinji.okumura@softfront.jp>
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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Gao,

This is a figure I've assumed.

    A                               B
    |                               |
  s0|re-INV (offer1)                |s0
    |------------------------------>|
    |               18x-rel(answer1)|
    |<------------------------------|
  s1|                        non-2xx|s1
    |<------------------------------|<-+
  s0|ACK                            |s0|
    |----------->                   |  |
    |UPD(offer0)                    |  |
    |==============================>|* | reject UPDATE
    |                       491(UPD)|  |
    |<==============================|  |
    |                    ACK        |  |
    |                    ---------->|<-+
    |                               |
                Figure 2

>Hi,
>
>OKUMURA Shinji <shinji.okumura@softfront.jp>
>> Gao,
>> 
>> gao.yang2@zte.com.cn
>> Tue, 25 May 2010 14:15:10 +0800
>> >Hi,
>> >
>> >> >> I agree that it doesn't forbid sending it.  It allows it and 
>> >> >> mandates the UAS return 491 or 500 (depending upon direction).
>> >> >
>> >> >It mandates the UAS send 4xx/5xx to UPDATE only when the UAS
>> >> >send 4xx/5xx to the Re-INVITE and the UAS get the UPDATE before
>> >> >ACK of 4xx/5xx(Re-INVITE).
>> >> 
>> >> As you know, ACK to non-2xx is hop-by-hop.
>> >> There is little benefit to reject such an UPDATE.
>> >
>> >UAS send 491 for the UPDATE(in the flow) to *tell* UAC that in it's
>> >(UAS) 
>> >view, there's a glare case.
>> 
>> But this UPDATE is sent in order to synchronize the common view.
>> If UAS return 2xx, then the situation goes into nonglare.
>
>I guess I have a inverse point here. By this draft's clarification, 
>there's no inconsistent session state for the non-successful Re-INVITE. 
>So, why UAC send a UPDATE to synchronize the common view?

In 3.4.  UAC Behavior,
  "In order to cope with these race condition
   situations, a UAC that receives an error response to a re-INVITE for
   which changes have been already executed SHOULD generate a new re-
   INVITE or UPDATE request in order to make sure that both UAs have a
   common view of the state of the dialog and the session."

Under the above descriptions, this UAC generates a new UPDATE.

>While if UAS return 2xx(for the UPDATE), the situation would get into 
>glare(UAS has no way to figure out this UPDATE is in-scope of Re-INVITE or 
>not). This is relating to atomic of session state. I'd like to emphasize 
>that keeping Re-INVITE atomic, except for target refresh, is the main idea 
>in this draft.

This issue has nothing to do with the "atomic of session state".

>> Probably another UPDATE will be sent, but this UPDATE
>> transaction is supposed to be avoided.
>> 
>> At the very start, the UAS rejects a reINVITE does not care
>> about this text. Is it necessary to let such a UAS follow a
>> new rule?
>
>These days, you guys showed a lot of use cases. I don't remember the 
>details. Could you say more about this use case.

Please look at the original mail of mine, and re-comment it.
You seem to be confused a little.

>> >When UAS get the ACK then UPDATE, it still has no precise criterion for 
>> >whether the UAC gets non-2xx(Re-INVITE) first then send the UPDATE or not. 
>> >Keeping rejecting the UPDATE is not a reasonable way here. So, the draft 
>> >suggest the way, as UAS can send 2xx to the UPDATE.
>> 
>> It is a conventional handling and enough for UAS.
>> 
>> >When UAC get UAS's 4xx of UPDATE, the session state is clear for it. And 
>> >UAC would only send a synchronizing Re-INVITE/UPDATE when it is a glare 
>> >case in the UAC's view.
>> 
>> UAS' behavior does not affect UAC' behavior directly.
>> UAC will do the needful regardless of whether UAS rejects
>> a reINVITE.
>> 
>> >IMO, this way is better than the way we might once use: allowing B send
>> >2xx to UPDATE, and both A and B *see* glare, and all send Re-INVITE/UPDATE 
>> >to refresh session state by each's view.
>> 
>> The new rule for UAS does not seem to improve the situation.
>> The new rule for UAC seems to be just enough to synchronize
>> the view.
>
>The key merit for the new rule is make both UAC and UAS clear in which 
>situation is it's responsibility to handle glare case, and there's only 
>one UA need to do something. Both sides want to refresh session state by 
>each's view is not a elegant way.

The crux of the matter is that a UAS can not recognize own glare
situation anytime.
It is UAC's duty to synchronize a common view.

Regards,
Shinji

>Thanks,
>
>Gao

From christer.holmberg@ericsson.com  Tue May 25 11:51:18 2010
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Thread-Topic: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without	SDP
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without	SDP
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Hi,=20

>Thanks for the clarification. So the changes do not prohibit the use of UP=
DATE after only a non-reliable provisional response, provided the UPDATE re=
quest does not contain an offer - correct? If it does contain an offer it i=
s subject=20
>to the glare rules.

Maybe I didn't understand, but what is the difference if you receive a reli=
able or unreliable response (without SDP)?

Regards,

Christer


> -----Original Message-----
> From: Brett Tate [mailto:brett@broadsoft.com]
> Sent: 24 May 2010 12:55
> To: Elwell, John; sipcore@ietf.org
> Subject: RE: [sipcore] draft-ietf-sipcore-reinvite-04: glare and=20
> UPDATE without SDP
>=20
> > > Draft-ietf-sipcore-reinvite-04 sections 3.5 and 4.5 indicate that=20
> > > an UPDATE without SDP should be treated (from a glare
> > > perspective) as though it contained an offer SDP.  However I=20
> > > currently find the definition of "glare" and the expected behavior=20
> > > to be ambiguous.
> > >
> > > What is the precise meaning of glare?  I ask mainly because some=20
> > > vendors send a 491 when an RFC indicates to send a 500.
> > >
> > > If INVITE contains offer SDP and 18x was not reliable, how SHOULD=20
> > > the called party respond if UPDATE without SDP is received from=20
> > > caller: 491, 500, or process as though no glare?
> > >
> > > If INVITE contains offer SDP and 18x was not reliable, how SHOULD=20
> > > the caller respond if UPDATE without SDP is received from called=20
> > > party: 491, 500, or process as though no glare?
> >
> > [JRE] I believe these two situations are both valid and should not=20
> > result in a 4xx/5xx/6xx response unless something else is wrong with=20
> > the request.
>=20
> That was true before draft-ietf-sipcore-reinvite.  However it looks=20
> like the following snippets make it no longer true.
>=20
> Draft-ietf-sipcore-reinvite section 3.5:
>=20
> "An UPDATE request without an offer can change dialog parameters.  So =20
> can a non-2xx final response to a re-INVITE request or a 2xx response =20
> to an INVITE request (re-INVITE or initial INVITE).  However, there =20
> are no rules to avoid a collision between an offerless UPDATE request =20
> and a final response to an INVITE request.  The rules in Section 4.5, =20
> which are given in the context of target refreshes, cover these types =20
> of collisions as well.  Therefore, there is no need to specify =20
> further rules here."
>=20
> Draft-ietf-sipcore-reinvite section 4.5:
>=20
> "When checking for glare situations, UAs MUST treat every UPDATE =20
> request as if it contained an offer."
>=20
> RFC3311 section 5.2:
>=20
> "If an UPDATE is received that contains an offer, and the UAS has =20
> generated an offer (in an UPDATE, PRACK or INVITE) to which it has =20
> not yet received an answer, the UAS MUST reject the UPDATE with a 491 =20
> response.  Similarly, if an UPDATE is received that contains an =20
> offer, and the UAS has received an offer (in an UPDATE, PRACK, or
>  INVITE) to which it has not yet generated an answer, the UAS MUST =20
> reject the UPDATE with a 500 response, and MUST include a Retry-After =20
> header field with a randomly chosen value between 0 and 10 seconds."
>=20
>=20
_______________________________________________
sipcore mailing list
sipcore@ietf.org
https://www.ietf.org/mailman/listinfo/sipcore

From john.elwell@siemens-enterprise.com  Tue May 25 12:11:41 2010
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Brett Tate <brett@broadsoft.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Tue, 25 May 2010 21:11:29 +0200
Thread-Topic: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without	SDP
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without	SDP
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> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: 25 May 2010 19:51
> To: Elwell, John; Brett Tate; sipcore@ietf.org
> Subject: RE: [sipcore] draft-ietf-sipcore-reinvite-04: glare=20
> and UPDATE without SDP
>=20
>=20
> Hi,=20
>=20
> >Thanks for the clarification. So the changes do not prohibit=20
> the use of UPDATE after only a non-reliable provisional=20
> response, provided the UPDATE request does not contain an=20
> offer - correct? If it does contain an offer it is subject=20
> >to the glare rules.
>=20
> Maybe I didn't understand, but what is the difference if you=20
> receive a reliable or unreliable response (without SDP)?
[JRE] If only an unreliable provisional response has been received , an UPD=
ATE is not allowed to contain SDP, so treating it as if it contained SDP is=
 rather odd. The following sequence:
(re-)INVITE----->
<-----18x (unreliable)
----->UPDATE (no SDP)
is valid in accordance with RFC 3311, I believe. What seems to be suggested=
 by the draft is that in some circumstances the UAS might treat the receive=
d UPDATE as a glare. It is not clear to me under what conditions it might b=
e treated as a glare, but in any case, that would seem to be a change to RF=
C 3311 for those particular conditions.

John

>=20
> Regards,
>=20
> Christer
>=20
>=20
> > -----Original Message-----
> > From: Brett Tate [mailto:brett@broadsoft.com]
> > Sent: 24 May 2010 12:55
> > To: Elwell, John; sipcore@ietf.org
> > Subject: RE: [sipcore] draft-ietf-sipcore-reinvite-04: glare and=20
> > UPDATE without SDP
> >=20
> > > > Draft-ietf-sipcore-reinvite-04 sections 3.5 and 4.5=20
> indicate that=20
> > > > an UPDATE without SDP should be treated (from a glare
> > > > perspective) as though it contained an offer SDP.  However I=20
> > > > currently find the definition of "glare" and the=20
> expected behavior=20
> > > > to be ambiguous.
> > > >
> > > > What is the precise meaning of glare?  I ask mainly=20
> because some=20
> > > > vendors send a 491 when an RFC indicates to send a 500.
> > > >
> > > > If INVITE contains offer SDP and 18x was not reliable,=20
> how SHOULD=20
> > > > the called party respond if UPDATE without SDP is received from=20
> > > > caller: 491, 500, or process as though no glare?
> > > >
> > > > If INVITE contains offer SDP and 18x was not reliable,=20
> how SHOULD=20
> > > > the caller respond if UPDATE without SDP is received=20
> from called=20
> > > > party: 491, 500, or process as though no glare?
> > >
> > > [JRE] I believe these two situations are both valid and=20
> should not=20
> > > result in a 4xx/5xx/6xx response unless something else is=20
> wrong with=20
> > > the request.
> >=20
> > That was true before draft-ietf-sipcore-reinvite.  However it looks=20
> > like the following snippets make it no longer true.
> >=20
> > Draft-ietf-sipcore-reinvite section 3.5:
> >=20
> > "An UPDATE request without an offer can change dialog=20
> parameters.  So =20
> > can a non-2xx final response to a re-INVITE request or a=20
> 2xx response =20
> > to an INVITE request (re-INVITE or initial INVITE). =20
> However, there =20
> > are no rules to avoid a collision between an offerless=20
> UPDATE request =20
> > and a final response to an INVITE request.  The rules in=20
> Section 4.5, =20
> > which are given in the context of target refreshes, cover=20
> these types =20
> > of collisions as well.  Therefore, there is no need to specify =20
> > further rules here."
> >=20
> > Draft-ietf-sipcore-reinvite section 4.5:
> >=20
> > "When checking for glare situations, UAs MUST treat every UPDATE =20
> > request as if it contained an offer."
> >=20
> > RFC3311 section 5.2:
> >=20
> > "If an UPDATE is received that contains an offer, and the UAS has =20
> > generated an offer (in an UPDATE, PRACK or INVITE) to which it has =20
> > not yet received an answer, the UAS MUST reject the UPDATE=20
> with a 491 =20
> > response.  Similarly, if an UPDATE is received that contains an =20
> > offer, and the UAS has received an offer (in an UPDATE, PRACK, or
> >  INVITE) to which it has not yet generated an answer, the UAS MUST =20
> > reject the UPDATE with a 500 response, and MUST include a=20
> Retry-After =20
> > header field with a randomly chosen value between 0 and 10 seconds."
> >=20
> >=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =

From christer.holmberg@ericsson.com  Tue May 25 12:58:47 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "'Elwell, John'" <john.elwell@siemens-enterprise.com>, Brett Tate <brett@broadsoft.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Tue, 25 May 2010 21:58:35 +0200
Thread-Topic: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without	SDP
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without	SDP
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Hi,=20

>>Thanks for the clarification. So the changes do not prohibit
>>the use of UPDATE after only a non-reliable provisional response,=20
>>provided the UPDATE request does not contain an offer - correct? If it=20
>>does contain an offer it is subject
>>to the glare rules.
>=20
>Maybe I didn't understand, but what is the difference if you receive a rel=
iable or unreliable response (without SDP)?
>
>[JRE] If only an unreliable provisional response has been received , an UP=
DATE is not allowed to contain SDP, so treating it as if it contained SDP i=
s rather odd.

But, how is it less "odd" than treating an UPDATE as it it contained SDP wh=
en you have received a reliable response (without SDP)?=20

>The following sequence:
>(re-)INVITE----->
><-----18x (unreliable)
>----->UPDATE (no SDP)
>is valid in accordance with RFC 3311, I believe.
>
>What seems to be suggested by the draft is that in some circumstances the =
UAS might treat the received UPDATE as a glare. It is not clear to me under=
 what conditions=20
>it might be treated as a glare, but in any case, that would seem to be a c=
hange to RFC 3311 for those particular conditions.

I will have to try to refresh my memory, because I can't see how it would b=
e a glare from an offer/answer perspective. Could it be related to target r=
efresh?

Regards,

Christer



>=20
> Regards,
>=20
> Christer
>=20
>=20
> > -----Original Message-----
> > From: Brett Tate [mailto:brett@broadsoft.com]
> > Sent: 24 May 2010 12:55
> > To: Elwell, John; sipcore@ietf.org
> > Subject: RE: [sipcore] draft-ietf-sipcore-reinvite-04: glare and=20
> > UPDATE without SDP
> >=20
> > > > Draft-ietf-sipcore-reinvite-04 sections 3.5 and 4.5
> indicate that
> > > > an UPDATE without SDP should be treated (from a glare
> > > > perspective) as though it contained an offer SDP.  However I=20
> > > > currently find the definition of "glare" and the
> expected behavior
> > > > to be ambiguous.
> > > >
> > > > What is the precise meaning of glare?  I ask mainly
> because some
> > > > vendors send a 491 when an RFC indicates to send a 500.
> > > >
> > > > If INVITE contains offer SDP and 18x was not reliable,
> how SHOULD
> > > > the called party respond if UPDATE without SDP is received from
> > > > caller: 491, 500, or process as though no glare?
> > > >
> > > > If INVITE contains offer SDP and 18x was not reliable,
> how SHOULD
> > > > the caller respond if UPDATE without SDP is received
> from called
> > > > party: 491, 500, or process as though no glare?
> > >
> > > [JRE] I believe these two situations are both valid and
> should not
> > > result in a 4xx/5xx/6xx response unless something else is
> wrong with
> > > the request.
> >=20
> > That was true before draft-ietf-sipcore-reinvite.  However it looks=20
> > like the following snippets make it no longer true.
> >=20
> > Draft-ietf-sipcore-reinvite section 3.5:
> >=20
> > "An UPDATE request without an offer can change dialog
> parameters.  So
> > can a non-2xx final response to a re-INVITE request or a
> 2xx response
> > to an INVITE request (re-INVITE or initial INVITE). =20
> However, there
> > are no rules to avoid a collision between an offerless
> UPDATE request
> > and a final response to an INVITE request.  The rules in
> Section 4.5,
> > which are given in the context of target refreshes, cover
> these types
> > of collisions as well.  Therefore, there is no need to specify=20
> > further rules here."
> >=20
> > Draft-ietf-sipcore-reinvite section 4.5:
> >=20
> > "When checking for glare situations, UAs MUST treat every UPDATE=20
> > request as if it contained an offer."
> >=20
> > RFC3311 section 5.2:
> >=20
> > "If an UPDATE is received that contains an offer, and the UAS has=20
> > generated an offer (in an UPDATE, PRACK or INVITE) to which it has=20
> > not yet received an answer, the UAS MUST reject the UPDATE
> with a 491
> > response.  Similarly, if an UPDATE is received that contains an=20
> > offer, and the UAS has received an offer (in an UPDATE, PRACK, or
> >  INVITE) to which it has not yet generated an answer, the UAS MUST=20
> > reject the UPDATE with a 500 response, and MUST include a
> Retry-After
> > header field with a randomly chosen value between 0 and 10 seconds."
> >=20
> >=20
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> =

From christer.holmberg@ericsson.com  Tue May 25 13:19:15 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "'Elwell, John'" <john.elwell@siemens-enterprise.com>, Brett Tate <brett@broadsoft.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Tue, 25 May 2010 22:18:34 +0200
Thread-Topic: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without	SDP
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without	SDP
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Hi,

So, please note that the text in section 4.5 is about target refreshes - no=
t offer/answers.

So, if you have received an unreliable response, I guess you could send an =
UPDATE. But, why would you want to? You cannot include SDP, and I don't bel=
ieve you can include updated target information.

But, if you have received an reliable response, you CAN send an UPDATE to u=
pdate the target - and so can the remote party. And, then you may end up wi=
th the glare situation described in section 4.5.

So, I believe that is the reason for the "treat the UPDATE as if it contain=
ed SDP" text, because in that case the UPDATE would be rejected (since you =
don't know if the remote end replied to your UPDATE before it sent its own =
UPDATE, or vice verse).

Brett said that the procedure "forces" the UAS to include SDP in the reliab=
le response, but I can't find any text indicating that...?

Regards,

Christer



=20

-----Original Message-----
From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behalf =
Of Christer Holmberg
Sent: 25. toukokuuta 2010 22:59
To: 'Elwell, John'; Brett Tate; sipcore@ietf.org
Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE wit=
hout SDP


Hi,=20

>>Thanks for the clarification. So the changes do not prohibit the use=20
>>of UPDATE after only a non-reliable provisional response, provided the=20
>>UPDATE request does not contain an offer - correct? If it does contain=20
>>an offer it is subject to the glare rules.
>=20
>Maybe I didn't understand, but what is the difference if you receive a rel=
iable or unreliable response (without SDP)?
>
>[JRE] If only an unreliable provisional response has been received , an UP=
DATE is not allowed to contain SDP, so treating it as if it contained SDP i=
s rather odd.

But, how is it less "odd" than treating an UPDATE as it it contained SDP wh=
en you have received a reliable response (without SDP)?=20

>The following sequence:
>(re-)INVITE----->
><-----18x (unreliable)
>----->UPDATE (no SDP)
>is valid in accordance with RFC 3311, I believe.
>
>What seems to be suggested by the draft is that in some circumstances=20
>the UAS might treat the received UPDATE as a glare. It is not clear to me =
under what conditions it might be treated as a glare, but in any case, that=
 would seem to be a change to RFC 3311 for those particular conditions.

I will have to try to refresh my memory, because I can't see how it would b=
e a glare from an offer/answer perspective. Could it be related to target r=
efresh?

Regards,

Christer



>=20
> Regards,
>=20
> Christer
>=20
>=20
> > -----Original Message-----
> > From: Brett Tate [mailto:brett@broadsoft.com]
> > Sent: 24 May 2010 12:55
> > To: Elwell, John; sipcore@ietf.org
> > Subject: RE: [sipcore] draft-ietf-sipcore-reinvite-04: glare and=20
> > UPDATE without SDP
> >=20
> > > > Draft-ietf-sipcore-reinvite-04 sections 3.5 and 4.5
> indicate that
> > > > an UPDATE without SDP should be treated (from a glare
> > > > perspective) as though it contained an offer SDP.  However I=20
> > > > currently find the definition of "glare" and the
> expected behavior
> > > > to be ambiguous.
> > > >
> > > > What is the precise meaning of glare?  I ask mainly
> because some
> > > > vendors send a 491 when an RFC indicates to send a 500.
> > > >
> > > > If INVITE contains offer SDP and 18x was not reliable,
> how SHOULD
> > > > the called party respond if UPDATE without SDP is received from
> > > > caller: 491, 500, or process as though no glare?
> > > >
> > > > If INVITE contains offer SDP and 18x was not reliable,
> how SHOULD
> > > > the caller respond if UPDATE without SDP is received
> from called
> > > > party: 491, 500, or process as though no glare?
> > >
> > > [JRE] I believe these two situations are both valid and
> should not
> > > result in a 4xx/5xx/6xx response unless something else is
> wrong with
> > > the request.
> >=20
> > That was true before draft-ietf-sipcore-reinvite.  However it looks=20
> > like the following snippets make it no longer true.
> >=20
> > Draft-ietf-sipcore-reinvite section 3.5:
> >=20
> > "An UPDATE request without an offer can change dialog
> parameters.  So
> > can a non-2xx final response to a re-INVITE request or a
> 2xx response
> > to an INVITE request (re-INVITE or initial INVITE). =20
> However, there
> > are no rules to avoid a collision between an offerless
> UPDATE request
> > and a final response to an INVITE request.  The rules in
> Section 4.5,
> > which are given in the context of target refreshes, cover
> these types
> > of collisions as well.  Therefore, there is no need to specify=20
> > further rules here."
> >=20
> > Draft-ietf-sipcore-reinvite section 4.5:
> >=20
> > "When checking for glare situations, UAs MUST treat every UPDATE=20
> > request as if it contained an offer."
> >=20
> > RFC3311 section 5.2:
> >=20
> > "If an UPDATE is received that contains an offer, and the UAS has=20
> > generated an offer (in an UPDATE, PRACK or INVITE) to which it has=20
> > not yet received an answer, the UAS MUST reject the UPDATE
> with a 491
> > response.  Similarly, if an UPDATE is received that contains an=20
> > offer, and the UAS has received an offer (in an UPDATE, PRACK, or
> >  INVITE) to which it has not yet generated an answer, the UAS MUST=20
> > reject the UPDATE with a 500 response, and MUST include a
> Retry-After
> > header field with a randomly chosen value between 0 and 10 seconds."
> >=20
> >=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 john.elwell@siemens-enterprise.com  Tue May 25 23:35:34 2010
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To: Christer Holmberg <christer.holmberg@ericsson.com>, Brett Tate <brett@broadsoft.com>, "sipcore@ietf.org" <sipcore@ietf.org>
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Thread-Topic: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without	SDP
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=20

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: 25 May 2010 20:59
> To: Elwell, John; Brett Tate; sipcore@ietf.org
> Subject: RE: [sipcore] draft-ietf-sipcore-reinvite-04: glare=20
> and UPDATE without SDP
>=20
>=20
> Hi,=20
>=20
> >>Thanks for the clarification. So the changes do not prohibit
> >>the use of UPDATE after only a non-reliable provisional response,=20
> >>provided the UPDATE request does not contain an offer -=20
> correct? If it=20
> >>does contain an offer it is subject
> >>to the glare rules.
> >=20
> >Maybe I didn't understand, but what is the difference if you=20
> receive a reliable or unreliable response (without SDP)?
> >
> >[JRE] If only an unreliable provisional response has been=20
> received , an UPDATE is not allowed to contain SDP, so=20
> treating it as if it contained SDP is rather odd.
>=20
> But, how is it less "odd" than treating an UPDATE as it it=20
> contained SDP when you have received a reliable response=20
> (without SDP)?=20
[JRE] You are right, a reliable response does not need to contain SDP if th=
ere was an offer in the INVITE request.

John

>=20
> >The following sequence:
> >(re-)INVITE----->
> ><-----18x (unreliable)
> >----->UPDATE (no SDP)
> >is valid in accordance with RFC 3311, I believe.
> >
> >What seems to be suggested by the draft is that in some=20
> circumstances the UAS might treat the received UPDATE as a=20
> glare. It is not clear to me under what conditions=20
> >it might be treated as a glare, but in any case, that would=20
> seem to be a change to RFC 3311 for those particular conditions.
>=20
> I will have to try to refresh my memory, because I can't see=20
> how it would be a glare from an offer/answer perspective.=20
> Could it be related to target refresh?
>=20
> Regards,
>=20
> Christer
>=20
>=20
>=20
> >=20
> > Regards,
> >=20
> > Christer
> >=20
> >=20
> > > -----Original Message-----
> > > From: Brett Tate [mailto:brett@broadsoft.com]
> > > Sent: 24 May 2010 12:55
> > > To: Elwell, John; sipcore@ietf.org
> > > Subject: RE: [sipcore] draft-ietf-sipcore-reinvite-04: glare and=20
> > > UPDATE without SDP
> > >=20
> > > > > Draft-ietf-sipcore-reinvite-04 sections 3.5 and 4.5
> > indicate that
> > > > > an UPDATE without SDP should be treated (from a glare
> > > > > perspective) as though it contained an offer SDP.  However I=20
> > > > > currently find the definition of "glare" and the
> > expected behavior
> > > > > to be ambiguous.
> > > > >
> > > > > What is the precise meaning of glare?  I ask mainly
> > because some
> > > > > vendors send a 491 when an RFC indicates to send a 500.
> > > > >
> > > > > If INVITE contains offer SDP and 18x was not reliable,
> > how SHOULD
> > > > > the called party respond if UPDATE without SDP is=20
> received from
> > > > > caller: 491, 500, or process as though no glare?
> > > > >
> > > > > If INVITE contains offer SDP and 18x was not reliable,
> > how SHOULD
> > > > > the caller respond if UPDATE without SDP is received
> > from called
> > > > > party: 491, 500, or process as though no glare?
> > > >
> > > > [JRE] I believe these two situations are both valid and
> > should not
> > > > result in a 4xx/5xx/6xx response unless something else is
> > wrong with
> > > > the request.
> > >=20
> > > That was true before draft-ietf-sipcore-reinvite. =20
> However it looks=20
> > > like the following snippets make it no longer true.
> > >=20
> > > Draft-ietf-sipcore-reinvite section 3.5:
> > >=20
> > > "An UPDATE request without an offer can change dialog
> > parameters.  So
> > > can a non-2xx final response to a re-INVITE request or a
> > 2xx response
> > > to an INVITE request (re-INVITE or initial INVITE). =20
> > However, there
> > > are no rules to avoid a collision between an offerless
> > UPDATE request
> > > and a final response to an INVITE request.  The rules in
> > Section 4.5,
> > > which are given in the context of target refreshes, cover
> > these types
> > > of collisions as well.  Therefore, there is no need to specify=20
> > > further rules here."
> > >=20
> > > Draft-ietf-sipcore-reinvite section 4.5:
> > >=20
> > > "When checking for glare situations, UAs MUST treat every UPDATE=20
> > > request as if it contained an offer."
> > >=20
> > > RFC3311 section 5.2:
> > >=20
> > > "If an UPDATE is received that contains an offer, and the UAS has=20
> > > generated an offer (in an UPDATE, PRACK or INVITE) to=20
> which it has=20
> > > not yet received an answer, the UAS MUST reject the UPDATE
> > with a 491
> > > response.  Similarly, if an UPDATE is received that contains an=20
> > > offer, and the UAS has received an offer (in an UPDATE, PRACK, or
> > >  INVITE) to which it has not yet generated an answer, the=20
> UAS MUST=20
> > > reject the UPDATE with a 500 response, and MUST include a
> > Retry-After
> > > header field with a randomly chosen value between 0 and=20
> 10 seconds."
> > >=20
> > >=20
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
> > =

From brett@broadsoft.com  Wed May 26 05:05:43 2010
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From: Brett Tate <brett@broadsoft.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Wed, 26 May 2010 05:04:28 -0700
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> So, please note that the text in section 4.5 is about target=20
> refreshes - not offer/answers.

Section 3.5 indicates that section 4.5 isn't just about target refreshes.

Section 4.5's fix explicitly uses offer/answer rules.  And as far as I know=
, it intentionally or unintentionally updates rfc3311 since all UPDATEs MUS=
T now be treated like they contained an offer.  If my understanding is not =
true, please explain how the following quote is to be interpreted.  Gao and=
 I appear to interpreting it (or when it applies) differently.

Draft-ietf-sipcore-reinvite section 4.5:

"When checking for glare situations, UAs MUST treat every UPDATE
 request as if it contained an offer."


> Brett said that the procedure "forces" the UAS to include SDP=20
> in the reliable response, but I can't find any text indicating=20
> that...?

I never used the word "forces".  And some may have thought my questions wer=
e associated with re-INVITE; my questions were/are associated with INVITE.=
=20

The following appears to require returning 491 or 500 to UPDATE without off=
er SDP when the INVITE's offer SDP has not been answered.  If my understand=
ing is incorrect, please explain how the normative text does not require it=
.

Draft-ietf-sipcore-reinvite section 3.5:

"An UPDATE request without an offer can change dialog parameters.  So
 can a non-2xx final response to a re-INVITE request or a 2xx response
 to an INVITE request (re-INVITE or initial INVITE).  However, there
 are no rules to avoid a collision between an offerless UPDATE request
 and a final response to an INVITE request.  The rules in Section 4.5,
 which are given in the context of target refreshes, cover these types
 of collisions as well.  Therefore, there is no need to specify
 further rules here."

Draft-ietf-sipcore-reinvite section 4.5:

"When checking for glare situations, UAs MUST treat every UPDATE
 request as if it contained an offer."

RFC3311 section 5.2:

"If an UPDATE is received that contains an offer, and the UAS has
 generated an offer (in an UPDATE, PRACK or INVITE) to which it has
 not yet received an answer, the UAS MUST reject the UPDATE with a 491
 response.  Similarly, if an UPDATE is received that contains an
 offer, and the UAS has received an offer (in an UPDATE, PRACK, or
 INVITE) to which it has not yet generated an answer, the UAS MUST
 reject the UPDATE with a 500 response, and MUST include a Retry-After
 header field with a randomly chosen value between 0 and 10 seconds."

Thanks,
Brett

From brett@broadsoft.com  Wed May 26 05:47:16 2010
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From: Brett Tate <brett@broadsoft.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "'Elwell, John'" <john.elwell@siemens-enterprise.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Wed, 26 May 2010 05:46:04 -0700
Thread-Topic: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without	SDP
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without	SDP
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> So, if you have received an unreliable response, I guess you=20
> could send an UPDATE. But, why would you want to? You cannot=20
> include SDP, and I don't believe you can include updated=20
> target information.

Prior to draft-ietf-sipcore-reinvite, you could use UPDATE without SDP to u=
pdate/communicate things without being confined to the current offer/answer=
 state.  For example, rfc5876 section 3.2 discusses communicating P-Asserte=
d-Identity.  It can no longer be successfully communicated within UPDATE ov=
er an early dialog (resulting in 2xx response) unless the called party has =
sent the answer for the INVITE's offer SDP.


> But, if you have received an reliable response, you CAN send an UPDATE
> to update the target - and so can the remote party. And, then you may
> end up with the glare situation described in section 4.5.

That isn't the only glare situation and it doesn't just apply to changing t=
arget.  As mentioned above, attempting to communicate P-Asserted-Identity w=
ill also lead to a 491 if the called party has not sent an answer SDP for I=
NVITE's offer SDP.


> So, I believe that is the reason for the "treat the UPDATE as if it
> contained SDP" text, because in that case the UPDATE would be rejected
> (since you don't know if the remote end replied to your UPDATE before
> it sent its own UPDATE, or vice verse).

The normative text does not limit it only that situation.

Draft-ietf-sipcore-reinvite-04 section 4.5:=20

"When checking for glare situations, UAs MUST treat every UPDATE
 request as if it contained an offer."

RFC3311 section 5.2:

"If an UPDATE is received that contains an offer, and the UAS has
 generated an offer (in an UPDATE, PRACK or INVITE) to which it has
 not yet received an answer, the UAS MUST reject the UPDATE with a 491
 response.  Similarly, if an UPDATE is received that contains an
 offer, and the UAS has received an offer (in an UPDATE, PRACK, or
 INVITE) to which it has not yet generated an answer, the UAS MUST
 reject the UPDATE with a 500 response, and MUST include a Retry-After
 header field with a randomly chosen value between 0 and 10 seconds."


From gao.yang2@zte.com.cn  Wed May 26 06:34:37 2010
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> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On Behal=
f
> Of Elwell, John
> Sent: Tuesday, May 25, 2010 4:34 AM
>=20
> Thanks, this addresses concerns I had with the previous version.

Same here.
=20
> Digging deeper into the issue with the Flow-Timer header field, I view it
> this way. The "keep" parameter is in the Via header field, which has one
> entry per hop over which the request passes. Therefore the "keep"
> parameter is clearly associated with a given hop. However, the Flow-Timer
> header field does not have the concept of one entry per hop. Therefore th=
e
> header field is assumed to apply only to the current hop, and has to be
> removed before forwarding a response over a hop on which keep-alive has
> not been requested, or needs to have its value modified before forwarding
> a response over a hop on which keep-alive has been requested. It seems od=
d
> that the two pieces of information used by this feature should be handled
> in two completely different ways.

Yes I had the same feeling when reading it - that it's odd/unnatural to do =
the Flow-Timer header in when it would be simpler to just make it the keep =
param's value.  But obviously Flow-Timer is used in Outbound, so in some se=
nse it also feels odd to make it different from that.

=20
> Another difference is that when a proxy receives a "keep" parameter in th=
e
> topmost Via header field entry, it adds "yes" as the value of that
> parameter when forwarding that Via header entry (which will no longer be
> the topmost entry and therefore will not be inspected by the next entity)=
.
> Therefore when the Via header field is returned in the response, it will
> already have the "yes" value in place when it crosses the hop to which th=
e
> entry applies. Therefore the proxy does not need to take special action
> concerning the Via header field when forwarding the response. In contrast=
,
> a proxy has to take special action with the Flow-Timer header field when
> forwarding a response, as described above. The way "keep" is handled seem=
s
> to take account of transaction-stateless proxies, whereas the way Flow-
> Timer is handle does not. I am not sure, however, whether this mechanism
> makes sense for use with transaction-stateless proxies anyway, so perhaps
> this latter point is moot.

We shouldn't disallow stateless proxies from doing it, but yeah it's not th=
e "normal" case.

Regardless, I'm not sure why the Proxy inserts the keep in the forwarded Re=
quest, instead of just in the response(s)?

-hadriel

From christer.holmberg@ericsson.com  Wed May 26 07:24:34 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>, SIPCORE <sipcore@ietf.org>
Date: Wed, 26 May 2010 16:24:21 +0200
Thread-Topic: Draft new version: draft-ietf-sipcore-keep-03
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Hi,

>>Thanks, this addresses concerns I had with the previous version.
>
>Same here.

Great!

>> Digging deeper into the issue with the Flow-Timer header field, I view i=
t
>> this way. The "keep" parameter is in the Via header field, which has one
>> entry per hop over which the request passes. Therefore the "keep"
>> parameter is clearly associated with a given hop. However, the Flow-Time=
r
>> header field does not have the concept of one entry per hop. Therefore t=
he
>> header field is assumed to apply only to the current hop, and has to be
>> removed before forwarding a response over a hop on which keep-alive has
>> not been requested, or needs to have its value modified before forwardin=
g
>> a response over a hop on which keep-alive has been requested. It seems o=
dd
>> that the two pieces of information used by this feature should be handle=
d
>> in two completely different ways.
>
>Yes I had the same feeling when reading it - that it's odd/unnatural to do=
 the Flow-Timer header in when it would be simpler to just make it the keep=
 param's value.  But obviously Flow-Timer is used in Outbound, so in some s=
ense it also feels odd to make it different from that.

So, we just need to pick one alternative :)

Keep is leading 1 - 0.

>> Another difference is that when a proxy receives a "keep" parameter in t=
he
>> topmost Via header field entry, it adds "yes" as the value of that
>> parameter when forwarding that Via header entry (which will no longer be
>> the topmost entry and therefore will not be inspected by the next entity=
).
>> Therefore when the Via header field is returned in the response, it will
>> already have the "yes" value in place when it crosses the hop to which t=
he
>> entry applies. Therefore the proxy does not need to take special action
>> concerning the Via header field when forwarding the response. In contras=
t,
>> a proxy has to take special action with the Flow-Timer header field when
>> forwarding a response, as described above. The way "keep" is handled see=
ms
>> to take account of transaction-stateless proxies, whereas the way Flow-
>> Timer is handle does not. I am not sure, however, whether this mechanism
>> makes sense for use with transaction-stateless proxies anyway, so perhap=
s
>> this latter point is moot.
>
>We shouldn't disallow stateless proxies from doing it, but yeah it's not t=
he "normal" case.
>
>Regardless, I'm not sure why the Proxy inserts the keep in the forwarded R=
equest, instead of just in the response(s)?

In case it wants to indicate to ITS adjacent upstream entity (maybe another=
 proxy) that it is willing to send-keep alives. There may not be many use-c=
ases (someone mentioned a PBX connection point acting as a proxy), but ther=
e is no reason to forbid it. UAs and proxies can use the same procedures.

Regards,

Christer=

From HKaplan@acmepacket.com  Wed May 26 07:40:30 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, SIPCORE <sipcore@ietf.org>
Date: Wed, 26 May 2010 10:40:18 -0400
Thread-Topic: Draft new version: draft-ietf-sipcore-keep-03
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Hi Christer,
I like this version a lot more. :)

Technical comments/questions:

1) Why does the Proxy insert the keep in the forwarded Request, vs. just th=
e response(s)?

2) You note that a keep=3Dyes in any response to a request means the next-h=
op can handle keep.  Does this mean a Proxy can send it in a 100 Trying?  I=
f so (and I assume it should be so), then the Proxy really needs to add the=
 "yes" to responses, not forwarded requests I think.

3) You use the phrase "within a registration" a lot.  I assume this means f=
or the lifetime of the Registration binding (as opposed to say the duration=
 of the REGISTER transaction, between the request and final response).  So =
does this mean that a REGISTER refresh (i.e., same call-id) does or does no=
t re-negotiate keep?  I think it must, since a real Proxy doesn't really kn=
ow whether a REGISTER is a refresh vs. a new one.

4) Section 4.2.1 states:
   "The scope of the negotiated keep-alives depends on what SIP request met=
hods are used for the keep-alive negotiation."
I don't think you want to make it method-specific in the draft, except for =
maybe REGISTER.  But other than REGISTER, any dialog-forming or in-dialog r=
equest affects the keep for the duration of the dialog.  Any out-of-dialog,=
 non-dialog-forming Request other than REGISTER would presumably have no im=
pact.  I mention this because an out-of-dialog MESSAGE or PUBLISH would hav=
e no bearing on keep I guess, but those same methods in-dialog could negoti=
ate keep (as I read the current text in the draft).


5) Section 4.2.1 states:
   "The sending and receiving of keep-alives can be negotiated when a
   registration or dialog is initiated, or later during the registration
   or dialog.  However, once a SIP entity has negotiated the sending of
   keep-alives within a registration or dialog, it can not re-negotiate
   the sending of keep-alives within the same registration or dialog.
   Likewise, once a SIP entity has indicated willingness to receive
   keep-alives within a registration or dialog, it MUST NOT indicate
   willingness to receive keep-alives in a response to a subsequent
   request within that registration or dialog."
Why is there that last sentence's restriction? =20

-hadriel

From HKaplan@acmepacket.com  Wed May 26 07:46:01 2010
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From: Hadriel Kaplan <HKaplan@acmepacket.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>, SIPCORE <sipcore@ietf.org>
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> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: Wednesday, May 26, 2010 10:24 AM
> To: Hadriel Kaplan; Elwell, John; SIPCORE
> >
> >Regardless, I'm not sure why the Proxy inserts the keep in the forwarded
> Request, instead of just in the response(s)?
>=20
> In case it wants to indicate to ITS adjacent upstream entity (maybe
> another proxy) that it is willing to send-keep alives. There may not be
> many use-cases (someone mentioned a PBX connection point acting as a
> proxy), but there is no reason to forbid it. UAs and proxies can use the
> same procedures.

Yeah I was worried you'd say that. :)

I think that's a dangerous assumption.  The Proxy may not be able to suppor=
t keep-alive on that Proxy-Proxy connection/side, while it can for the UA-P=
roxy side.  Believing some lower Via isn't safe.

What should really happen is the Proxy should add a keep param (without "ye=
s") in ITS Via header, so that the downstream Proxy getting it can add a "y=
es" to that Via, for that connection.  In other words, all the logic should=
 be about top-most Via's.

-hadriel

From christer.holmberg@ericsson.com  Wed May 26 08:19:08 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>, SIPCORE <sipcore@ietf.org>
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Subject: Re: [sipcore] Draft new version: draft-ietf-sipcore-keep-03
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Hi,

>>>Regardless, I'm not sure why the Proxy inserts the keep in the forwarded
>>Request, instead of just in the response(s)?
>
>>In case it wants to indicate to ITS adjacent upstream entity (maybe
>>another proxy) that it is willing to send-keep alives. There may not be
>>many use-cases (someone mentioned a PBX connection point acting as a
>>proxy), but there is no reason to forbid it. UAs and proxies can use the
>>same procedures.
>
>Yeah I was worried you'd say that. :)
>
>I think that's a dangerous assumption.  The Proxy may not be able to suppo=
rt keep-alive on that Proxy-Proxy connection/side, while it can for the UA-=
Proxy side.  Believing some lower Via isn't safe.

I am not sure I understand.

If the proxy wants to receive keep-alives on the UA-Proxy side, it inserts =
a "yes" parameter value (or, the frequency, if we choose that alternative) =
in the keep paramter inserted by the UA. in its Via parameter

If the proxy wants to send keep-alives towards the next hop proxy, it inser=
ts a keep paramter in its own Via header.

So, there are two different procedures for indicating willingness to send a=
nd received.

>What should really happen is the Proxy should add a keep param (without "y=
es") in ITS Via header, so that the downstream Proxy getting it can add a "=
yes" to that Via, for that connection.  In other words, all the logic shoul=
d be about top-most Via's.

That is how it works. If you are willing to receive, you insert "yes" in th=
e Via of the previous hop (if it contains "keep", or course). If you are wi=
lling to send, you insert "keep" in your OWN Via.

Regards,

Christer=

From brett@broadsoft.com  Wed May 26 08:19:41 2010
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From: Brett Tate <brett@broadsoft.com>
To: "gao.yang2@zte.com.cn" <gao.yang2@zte.com.cn>
Date: Wed, 26 May 2010 08:18:26 -0700
Thread-Topic: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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You and I know where our disagreement is concerning what the draft mandates=
.  We are still waiting to hear for Paul, Christer, and Gonzalo concerning =
how to interpret the following quote.

"When checking for glare situations, UAs MUST treat every UPDATE request as=
 if it contained an offer."

> I guess the key for this divarication here is still about=20
> the different understanding of the semantics of "checking=20
> for glare situations". If it is only applied in happened=20
> glare situations, it would not be a problem.=20

How can you only apply "checking for glare situations" during a glare situa=
tion?

Checking for offer/answer glare is part of normal RFC3311 UPDATE processing=
.

RFC3311 section 5.2:

"If an UPDATE is received that contains an offer, and the UAS has
 generated an offer (in an UPDATE, PRACK or INVITE) to which it has
 not yet received an answer, the UAS MUST reject the UPDATE with a 491
 response.  Similarly, if an UPDATE is received that contains an
 offer, and the UAS has received an offer (in an UPDATE, PRACK, or
 INVITE) to which it has not yet generated an answer, the UAS MUST
 reject the UPDATE with a 500 response, and MUST include a Retry-After
 header field with a randomly chosen value between 0 and 10 seconds."

> To make the two understanding's merit and disadvantage clear,=20
> I'd like to say one disadvantage for the understanding you showed:=20
> If the rules is for all target refresh request without O/A, it=20
> really forbid some behaviors which is lawful once. No matter how=20
> much behavior would be affected, it would be a real and remarkable=20
> compatible interworking problem.=20

I agree.  That is why I'm trying to understand what the draft mandates or a=
ttempted to mandate.

> So, urgency here is about the understanding of the semantics of=20
> "checking for glare situations".=20

I agree.

> And I'd like to hear your comments about the cases mentioned under=20
> my understanding "checking for glare situations".

I don't know how to reconcile your interpretation with the RFC3311 section =
5.2 snippet checking for offer/answer glare (shown above).


From christer.holmberg@ericsson.com  Wed May 26 08:31:23 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>, SIPCORE <sipcore@ietf.org>
Date: Wed, 26 May 2010 17:31:07 +0200
Thread-Topic: Draft new version: draft-ietf-sipcore-keep-03
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Hi,

>Technical comments/questions:
>
>1) Why does the Proxy insert the keep in the forwarded Request, vs. just t=
he response(s)?

See my reply to your other mail.

The proxy inserts "keep" in itw own Via to indicate willingess to SEND keep=
-alives towards the adjacent upstream hop.

The proxy inserts "=3Dyes" in the Via of the adjacent downstream hop to ind=
iciate willingness to RECEIVE keep-alives from that hop.

>2) You note that a keep=3Dyes in any response to a request means the next-=
hop can handle keep.  Does this mean a Proxy can send it in a 100 Trying?  =
If so (and I assume it should be so), then the Proxy really needs to add th=
e "yes" to responses, not forwarded requests I think.

That's a good point. We can change that.

The only reason it currently adds "=3Dyes" to the request is because I thou=
ght it is alligned with the received parameter. There is really no technica=
l reason to do so, because it is the adjacent downstream entity that needs =
the information.

>3) You use the phrase "within a registration" a lot.  I assume this means =
for the lifetime of the Registration binding (as opposed to say the duratio=
n of the REGISTER transaction, between the request and final response).

Correct.

>So does this mean that a REGISTER refresh (i.e., same call-id) does or doe=
s not re-negotiate keep?  I think it must, since a real Proxy doesn't reall=
y know whether a REGISTER is a refresh vs. a new one.

That's a good point. I will think a little about it, but if you're right th=
e keep-alive would have to be re-negotiated in registration refreshes.=20

That should not be a big problem, though, as registration re-refreshes don'=
t happen that occur (especially since one will not have to use them for kee=
p-alives anymore).

>4) Section 4.2.1 states:
> "The scope of the negotiated keep-alives depends on what SIP request meth=
ods are used for the keep-alive negotiation."
>I don't think you want to make it method-specific in the draft, except for=
 maybe REGISTER.  But other than REGISTER, any dialog-forming or in-dialog =
request affects the keep for the duration of the dialog.

The text DO talk about initial requests for dialogs, and mid-dialog target =
refresh requests. It does not talk about methods for dialog keep-alives. So=
, the text you refer to probably needs to be modified. Maybe it should say =
"what type of SIP requests are used for...".

>Any out-of-dialog, non-dialog-forming Request other than REGISTER would pr=
esumably have no impact.  I mention this because an out-of-dialog MESSAGE o=
r PUBLISH would have no bearing on keep I guess, but those same methods in-=
dialog could negotiate keep (as I read the current text in the draft).

Correct. It would probably be good to mention that also.

>5) Section 4.2.1 states:
>   "The sending and receiving of keep-alives can be negotiated when a
>   registration or dialog is initiated, or later during the registration
>   or dialog.  However, once a SIP entity has negotiated the sending of
>   keep-alives within a registration or dialog, it can not re-negotiate
>   the sending of keep-alives within the same registration or dialog.
>   Likewise, once a SIP entity has indicated willingness to receive
>   keep-alives within a registration or dialog, it MUST NOT indicate
>   willingness to receive keep-alives in a response to a subsequent
>   request within that registration or dialog."
>   Why is there that last sentence's restriction?

Maybe we don't need the sentence, becuase it should not receive a keep para=
mter within the dialog in the first place.

Based on your previous comment about proxies not knowing whether a REGISTER=
 is a refresh or not, for registrations we probably need to say that it MUS=
T indicate willingness to receive also for registration refreshes.

Regards,

Christer=

From christer.holmberg@ericsson.com  Wed May 26 08:33:46 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Brett Tate <brett@broadsoft.com>, "gao.yang2@zte.com.cn" <gao.yang2@zte.com.cn>
Date: Wed, 26 May 2010 17:31:20 +0200
Thread-Topic: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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Hi,

Just to let you know, Gonzalo is out of office this and next week, with lim=
ited access to his e-mail, so I doubt you will get his opinion before he is=
 back.

I tried earlier to explain earlier how I understand the quote, but based on=
 the previous replies I need to do some more reading.

Regards,

Christer



________________________________________
From: Brett Tate [brett@broadsoft.com]
Sent: Wednesday, May 26, 2010 6:18 PM
To: gao.yang2@zte.com.cn
Cc: Christer Holmberg; sipcore@ietf.org
Subject: RE: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE wit=
hout SDP

You and I know where our disagreement is concerning what the draft mandates=
.  We are still waiting to hear for Paul, Christer, and Gonzalo concerning =
how to interpret the following quote.

"When checking for glare situations, UAs MUST treat every UPDATE request as=
 if it contained an offer."

> I guess the key for this divarication here is still about
> the different understanding of the semantics of "checking
> for glare situations". If it is only applied in happened
> glare situations, it would not be a problem.

How can you only apply "checking for glare situations" during a glare situa=
tion?

Checking for offer/answer glare is part of normal RFC3311 UPDATE processing=
.

RFC3311 section 5.2:

"If an UPDATE is received that contains an offer, and the UAS has
 generated an offer (in an UPDATE, PRACK or INVITE) to which it has
 not yet received an answer, the UAS MUST reject the UPDATE with a 491
 response.  Similarly, if an UPDATE is received that contains an
 offer, and the UAS has received an offer (in an UPDATE, PRACK, or
 INVITE) to which it has not yet generated an answer, the UAS MUST
 reject the UPDATE with a 500 response, and MUST include a Retry-After
 header field with a randomly chosen value between 0 and 10 seconds."

> To make the two understanding's merit and disadvantage clear,
> I'd like to say one disadvantage for the understanding you showed:
> If the rules is for all target refresh request without O/A, it
> really forbid some behaviors which is lawful once. No matter how
> much behavior would be affected, it would be a real and remarkable
> compatible interworking problem.

I agree.  That is why I'm trying to understand what the draft mandates or a=
ttempted to mandate.

> So, urgency here is about the understanding of the semantics of
> "checking for glare situations".

I agree.

> And I'd like to hear your comments about the cases mentioned under
> my understanding "checking for glare situations".

I don't know how to reconcile your interpretation with the RFC3311 section =
5.2 snippet checking for offer/answer glare (shown above).=

From HKaplan@acmepacket.com  Wed May 26 08:33:58 2010
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To: Christer Holmberg <christer.holmberg@ericsson.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>, SIPCORE <sipcore@ietf.org>
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Ahh, sorry for the confusion.  It occurs to me I didn't ask the question co=
rrectly the first time, which resulted in us talking about different things=
.

In the current draft, I believe that a Proxy inserts the "yes" value for th=
e keep of the Via it receives from the UA, in the *forwarded request* it se=
nds onwards.  My question is why is that done, instead of just inserting "y=
es" in the response(s).

In other words, why is this done:
     Alice                        P1                      REGISTRAR
       |                          |                           |
       |--- REGISTER------------->|                           |
       |    Via: UAC;keep         |                           |
       |                          |--- REGISTER-------------->|
       |                          |    Via: P1;keep           |
       |                          |    Via: UAC;keep=3Dyes      |
       |                          |                           |
       |                          |<-- 200 OK ----------------|
       |                          |    Via: P1;keep=3Dyes       |
       |                          |    Via: UAC;keep=3Dyes      |
       |<-- 200 OK ---------------|                           |
       |    Via: UAC;keep=3Dyes     |                           |
       |    Flow-Timer: 30        |                           |


Instead of this:
     Alice                        P1                      REGISTRAR
       |                          |                           |
       |--- REGISTER------------->|                           |
       |    Via: UAC;keep         |                           |
       |                          |--- REGISTER-------------->|
       |                          |    Via: P1;keep           |
       |                          |    Via: UAC;keep          |
       |                          |                           |
       |                          |<-- 200 OK ----------------|
       |                          |    Via: P1;keep=3Dyes       |
       |                          |    Via: UAC;keep          |
       |<-- 200 OK ---------------|                           |
       |    Via: UAC;keep=3Dyes     |                           |
       |    Flow-Timer: 30        |                           |


> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: Wednesday, May 26, 2010 11:15 AM
> To: Hadriel Kaplan; Elwell, John; SIPCORE
> Subject: RE: Draft new version: draft-ietf-sipcore-keep-03
>=20
> Hi,
>=20
> >>>Regardless, I'm not sure why the Proxy inserts the keep in the
> forwarded
> >>Request, instead of just in the response(s)?
> >
> >>In case it wants to indicate to ITS adjacent upstream entity (maybe
> >>another proxy) that it is willing to send-keep alives. There may not be
> >>many use-cases (someone mentioned a PBX connection point acting as a
> >>proxy), but there is no reason to forbid it. UAs and proxies can use th=
e
> >>same procedures.
> >
> >Yeah I was worried you'd say that. :)
> >
> >I think that's a dangerous assumption.  The Proxy may not be able to
> support keep-alive on that Proxy-Proxy connection/side, while it can for
> the UA-Proxy side.  Believing some lower Via isn't safe.
>=20
> I am not sure I understand.
>=20
> If the proxy wants to receive keep-alives on the UA-Proxy side, it insert=
s
> a "yes" parameter value (or, the frequency, if we choose that alternative=
)
> in the keep paramter inserted by the UA. in its Via parameter
>=20
> If the proxy wants to send keep-alives towards the next hop proxy, it
> inserts a keep paramter in its own Via header.
>=20
> So, there are two different procedures for indicating willingness to send
> and received.
>=20
> >What should really happen is the Proxy should add a keep param (without
> "yes") in ITS Via header, so that the downstream Proxy getting it can add
> a "yes" to that Via, for that connection.  In other words, all the logic
> should be about top-most Via's.
>=20
> That is how it works. If you are willing to receive, you insert "yes" in
> the Via of the previous hop (if it contains "keep", or course). If you ar=
e
> willing to send, you insert "keep" in your OWN Via.
>=20
> Regards,
>=20
> Christer

From pkyzivat@cisco.com  Wed May 26 08:53:37 2010
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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Brett,

I'm aware. I got behind, and need to read through a whole bunch of 
messages carefully before I can comment meaningfully. Because of the 
backlog I now need some unbroken time to study all those, and haven't 
been able to steal it from my day job yet.

My general (tentative) feeling right now is that the requirement in 
-reinvite- to treat UPDATEs as if they contained an offer is 
problematic. It may be necessary to spell out explicitly the desired 
behavior rather than doing so indirectly like this.

	Thanks,
	Paul

Brett Tate wrote:
> You and I know where our disagreement is concerning what the draft mandates.  We are still waiting to hear for Paul, Christer, and Gonzalo concerning how to interpret the following quote.
> 
> "When checking for glare situations, UAs MUST treat every UPDATE request as if it contained an offer."
> 
>> I guess the key for this divarication here is still about 
>> the different understanding of the semantics of "checking 
>> for glare situations". If it is only applied in happened 
>> glare situations, it would not be a problem. 
> 
> How can you only apply "checking for glare situations" during a glare situation?
> 
> Checking for offer/answer glare is part of normal RFC3311 UPDATE processing.
> 
> RFC3311 section 5.2:
> 
> "If an UPDATE is received that contains an offer, and the UAS has
>  generated an offer (in an UPDATE, PRACK or INVITE) to which it has
>  not yet received an answer, the UAS MUST reject the UPDATE with a 491
>  response.  Similarly, if an UPDATE is received that contains an
>  offer, and the UAS has received an offer (in an UPDATE, PRACK, or
>  INVITE) to which it has not yet generated an answer, the UAS MUST
>  reject the UPDATE with a 500 response, and MUST include a Retry-After
>  header field with a randomly chosen value between 0 and 10 seconds."
> 
>> To make the two understanding's merit and disadvantage clear, 
>> I'd like to say one disadvantage for the understanding you showed: 
>> If the rules is for all target refresh request without O/A, it 
>> really forbid some behaviors which is lawful once. No matter how 
>> much behavior would be affected, it would be a real and remarkable 
>> compatible interworking problem. 
> 
> I agree.  That is why I'm trying to understand what the draft mandates or attempted to mandate.
> 
>> So, urgency here is about the understanding of the semantics of 
>> "checking for glare situations". 
> 
> I agree.
> 
>> And I'd like to hear your comments about the cases mentioned under 
>> my understanding "checking for glare situations".
> 
> I don't know how to reconcile your interpretation with the RFC3311 section 5.2 snippet checking for offer/answer glare (shown above).
> 
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 

From christer.holmberg@ericsson.com  Wed May 26 11:43:13 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Hadriel Kaplan <HKaplan@acmepacket.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>, SIPCORE <sipcore@ietf.org>
Date: Wed, 26 May 2010 20:38:54 +0200
Thread-Topic: Draft new version: draft-ietf-sipcore-keep-03
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Subject: Re: [sipcore] Draft new version: draft-ietf-sipcore-keep-03
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Hi,

>Ahh, sorry for the confusion.  It occurs to me I didn't ask the question c=
orrectly the first time, which resulted in us talking about different thing=
s.
>
>In the current draft, I believe that a Proxy inserts the "yes" value for t=
he keep of the Via it receives from the UA, in the *forwarded request* it s=
ends onwards.  My question is why is that done, instead of just inserting "=
yes" in the response(s).

As I said earlier, I can insert it in the response. It doesn't need to be i=
n the request. I just thought that since the proxy anyway might insert the =
received parameter in the request, it could as well insert "=3Dyes" at the =
same time.

But, if it confuses readers of the draft we can change it, and specify that=
 "=3Dyes" is inserted in the response. It would work even for transaction s=
tateless proxies, since the "keep" parameter of the UA will be in the respo=
nse.

Regards,

Christer



In other words, why is this done:
     Alice                        P1                      REGISTRAR
       |                          |                           |
       |--- REGISTER------------->|                           |
       |    Via: UAC;keep         |                           |
       |                          |--- REGISTER-------------->|
       |                          |    Via: P1;keep           |
       |                          |    Via: UAC;keep=3Dyes      |
       |                          |                           |
       |                          |<-- 200 OK ----------------|
       |                          |    Via: P1;keep=3Dyes       |
       |                          |    Via: UAC;keep=3Dyes      |
       |<-- 200 OK ---------------|                           |
       |    Via: UAC;keep=3Dyes     |                           |
       |    Flow-Timer: 30        |                           |


Instead of this:
     Alice                        P1                      REGISTRAR
       |                          |                           |
       |--- REGISTER------------->|                           |
       |    Via: UAC;keep         |                           |
       |                          |--- REGISTER-------------->|
       |                          |    Via: P1;keep           |
       |                          |    Via: UAC;keep          |
       |                          |                           |
       |                          |<-- 200 OK ----------------|
       |                          |    Via: P1;keep=3Dyes       |
       |                          |    Via: UAC;keep          |
       |<-- 200 OK ---------------|                           |
       |    Via: UAC;keep=3Dyes     |                           |
       |    Flow-Timer: 30        |                           |


> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: Wednesday, May 26, 2010 11:15 AM
> To: Hadriel Kaplan; Elwell, John; SIPCORE
> Subject: RE: Draft new version: draft-ietf-sipcore-keep-03
>
> Hi,
>
> >>>Regardless, I'm not sure why the Proxy inserts the keep in the
> forwarded
> >>Request, instead of just in the response(s)?
> >
> >>In case it wants to indicate to ITS adjacent upstream entity (maybe
> >>another proxy) that it is willing to send-keep alives. There may not be
> >>many use-cases (someone mentioned a PBX connection point acting as a
> >>proxy), but there is no reason to forbid it. UAs and proxies can use th=
e
> >>same procedures.
> >
> >Yeah I was worried you'd say that. :)
> >
> >I think that's a dangerous assumption.  The Proxy may not be able to
> support keep-alive on that Proxy-Proxy connection/side, while it can for
> the UA-Proxy side.  Believing some lower Via isn't safe.
>
> I am not sure I understand.
>
> If the proxy wants to receive keep-alives on the UA-Proxy side, it insert=
s
> a "yes" parameter value (or, the frequency, if we choose that alternative=
)
> in the keep paramter inserted by the UA. in its Via parameter
>
> If the proxy wants to send keep-alives towards the next hop proxy, it
> inserts a keep paramter in its own Via header.
>
> So, there are two different procedures for indicating willingness to send
> and received.
>
> >What should really happen is the Proxy should add a keep param (without
> "yes") in ITS Via header, so that the downstream Proxy getting it can add
> a "yes" to that Via, for that connection.  In other words, all the logic
> should be about top-most Via's.
>
> That is how it works. If you are willing to receive, you insert "yes" in
> the Via of the previous hop (if it contains "keep", or course). If you ar=
e
> willing to send, you insert "keep" in your OWN Via.
>
> Regards,
>
> Christer=

From christer.holmberg@ericsson.com  Wed May 26 12:07:39 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@cisco.com>, Brett Tate <brett@broadsoft.com>
Date: Wed, 26 May 2010 21:07:25 +0200
Thread-Topic: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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Hi,

Isn't what the text is trying to say the following:

"When you are about to send an UPDATE, or when you receive an UPDATE, you s=
hall apply the glare checking rules as if the UPDATE would contain an SDP o=
ffer."

So, it means you can only send an empty UPDATE if you in the same situation=
 were able to send an UPDATE with an offer.

And, it means that you do whatever you would do if you have sent an UPDATE =
with an offer, and a glare situation occurs.

I am not sure what is meant by "while being in a glare situation". I guess =
it means "while being in a situation where a glare could occur", or somethi=
ng like that.

But, these are just some unreliable provisional thoughts - I still need to =
do further reading of the document.

Regards,

Christer





________________________________________
From: sipcore-bounces@ietf.org [sipcore-bounces@ietf.org] On Behalf Of Paul=
 Kyzivat [pkyzivat@cisco.com]
Sent: Wednesday, May 26, 2010 6:53 PM
To: Brett Tate
Cc: gao.yang2@zte.com.cn; sipcore@ietf.org
Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE wit=
hout SDP

Brett,

I'm aware. I got behind, and need to read through a whole bunch of
messages carefully before I can comment meaningfully. Because of the
backlog I now need some unbroken time to study all those, and haven't
been able to steal it from my day job yet.

My general (tentative) feeling right now is that the requirement in
-reinvite- to treat UPDATEs as if they contained an offer is
problematic. It may be necessary to spell out explicitly the desired
behavior rather than doing so indirectly like this.

        Thanks,
        Paul

Brett Tate wrote:
> You and I know where our disagreement is concerning what the draft mandat=
es.  We are still waiting to hear for Paul, Christer, and Gonzalo concernin=
g how to interpret the following quote.
>
> "When checking for glare situations, UAs MUST treat every UPDATE request =
as if it contained an offer."
>
>> I guess the key for this divarication here is still about
>> the different understanding of the semantics of "checking
>> for glare situations". If it is only applied in happened
>> glare situations, it would not be a problem.
>
> How can you only apply "checking for glare situations" during a glare sit=
uation?
>
> Checking for offer/answer glare is part of normal RFC3311 UPDATE processi=
ng.
>
> RFC3311 section 5.2:
>
> "If an UPDATE is received that contains an offer, and the UAS has
>  generated an offer (in an UPDATE, PRACK or INVITE) to which it has
>  not yet received an answer, the UAS MUST reject the UPDATE with a 491
>  response.  Similarly, if an UPDATE is received that contains an
>  offer, and the UAS has received an offer (in an UPDATE, PRACK, or
>  INVITE) to which it has not yet generated an answer, the UAS MUST
>  reject the UPDATE with a 500 response, and MUST include a Retry-After
>  header field with a randomly chosen value between 0 and 10 seconds."
>
>> To make the two understanding's merit and disadvantage clear,
>> I'd like to say one disadvantage for the understanding you showed:
>> If the rules is for all target refresh request without O/A, it
>> really forbid some behaviors which is lawful once. No matter how
>> much behavior would be affected, it would be a real and remarkable
>> compatible interworking problem.
>
> I agree.  That is why I'm trying to understand what the draft mandates or=
 attempted to mandate.
>
>> So, urgency here is about the understanding of the semantics of
>> "checking for glare situations".
>
> I agree.
>
>> And I'd like to hear your comments about the cases mentioned under
>> my understanding "checking for glare situations".
>
> I don't know how to reconcile your interpretation with the RFC3311 sectio=
n 5.2 snippet checking for offer/answer glare (shown above).
>
> _______________________________________________
> 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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> How can you only apply "checking for glare situations" during a 
> glare situation?
> 
> Checking for offer/answer glare is part of normal RFC3311 UPDATE 
processing.

Making two analyzers in series as:

---> analyzer1 -----> analyzer2 ---->

analyzer1 using the rules in RFC3311, and if there's glares case under 
it's rule, send 491 or (500+retry) directly;
analyzer2 using the rules in the draft. 

analyzer2 also has two process:
1. Check whether it can do the ordering of 2xx(INVITE)and UPDATE. If it 
can do so, pass the UPDATE to the next step normally;
2. If it can not do the ordering, handle the UPDATE and 2xx(INVITE) as if 
it contains Offer and using O/A rules to reject.

The effected cases/siutation is only these cases/situation which are 
ambiguous before the draft.


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<br><tt><font size=2><br>
&gt; How can you only apply &quot;checking for glare situations&quot; during
a <br>
&gt; glare situation?<br>
&gt; <br>
&gt; Checking for offer/answer glare is part of normal RFC3311 UPDATE processing.<br>
<br>
Making two analyzers in series as:</font></tt>
<br>
<br><tt><font size=2>---&gt; analyzer1 -----&gt; analyzer2 ----&gt;</font></tt>
<br>
<br><tt><font size=2>analyzer1 using the rules in RFC3311, and if there's
glares case under it's rule, send 491 or (500+retry) directly;</font></tt>
<br><tt><font size=2>analyzer2 using the rules in the draft. </font></tt>
<br>
<br><tt><font size=2>analyzer2 also has two process:</font></tt>
<br><tt><font size=2>1. Check whether it can do the ordering of 2xx(INVITE)and
UPDATE. If it can do so, pass the UPDATE to the next step normally;</font></tt>
<br><tt><font size=2>2. If it can not do the ordering, handle the UPDATE
and 2xx(INVITE) as if it contains Offer and using O/A rules to reject.</font></tt>
<br>
<br><tt><font size=2>The effected cases/siutation is only these cases/situation
which are ambiguous before the draft.</font></tt>
<br><br><pre>
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From christer.holmberg@ericsson.com  Thu May 27 02:51:51 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "gao.yang2@zte.com.cn" <gao.yang2@zte.com.cn>, Brett Tate <brett@broadsoft.com>
Date: Thu, 27 May 2010 11:51:38 +0200
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Below are some questions/ideas on how I think we could solve most issues. S=
ome of the stuff shall be seen more as brain storming than actual proposal =
at this point.

I am fully aware that some of the ideas for sure may limit what is said in =
RFCs etc, so you don't need to tell me about that.




IDEA #1:  Offer/answer: Usage of SDP o=3D version parameter

I know this has been discussed at some point, but what is the reason we can=
't use the SDP o=3D version parameter to determine in which order the UAS h=
as sent SDPs?

I believe it would solve offer/answer glare situations where the UAC does n=
ot known in which order the UAS has sent SDPs.

That should not be controversial, and does not break any RFC etc.




IDEA #2:   Target update: Only outside re-INVITEs

Let's forbid target updates "inside" re-INVITEs. I really doubt there is a =
need for those while the UAs are doing something else with the re-INVITE.

That at least solves the glare situation where the remote target may be upd=
ated in a re-INVITE response and an UPDATE response.




IDEA #3:    UPDATE: Always use re-INVITE glare rules

Let's not allow more than one outstanding UPDATE. Ie if a UA has sent an UP=
DATE (with or without SDP), and receives another UPDATE request before the =
answer, it rejects it (same rule as for re-INVITE).


That would at least solve the problem where a UA sends an UPDATE, receives =
a 200 OK and a UPDATE request, both with different target information.


Ok, let the tomatoes fly...

...and let me know if there are any issues which the ideas above do NOT sol=
ve. I have some more ideas, but I am not sure they are needed so I'll them =
to myself for now :)


Regards,

Christer



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To: "gao.yang2@zte.com.cn" <gao.yang2@zte.com.cn>
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Thread-Topic: [sipcore] New deal [ was draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP]
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From john.elwell@siemens-enterprise.com  Thu May 27 05:29:47 2010
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To: Brett Tate <brett@broadsoft.com>, Christer Holmberg <christer.holmberg@ericsson.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Thu, 27 May 2010 14:29:35 +0200
Thread-Topic: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without	SDP
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without	SDP
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=20

> -----Original Message-----
> From: Brett Tate [mailto:brett@broadsoft.com]=20
> Sent: 26 May 2010 13:46
> To: Christer Holmberg; Elwell, John; sipcore@ietf.org
> Subject: RE: [sipcore] draft-ietf-sipcore-reinvite-04: glare=20
> and UPDATE without SDP
>=20
> > So, if you have received an unreliable response, I guess you=20
> > could send an UPDATE. But, why would you want to? You cannot=20
> > include SDP, and I don't believe you can include updated=20
> > target information.
>=20
> Prior to draft-ietf-sipcore-reinvite, you could use UPDATE=20
> without SDP to update/communicate things without being=20
> confined to the current offer/answer state.  For example,=20
> rfc5876 section 3.2 discusses communicating=20
> P-Asserted-Identity.  It can no longer be successfully=20
> communicated within UPDATE over an early dialog (resulting in=20
> 2xx response) unless the called party has sent the answer for=20
> the INVITE's offer SDP.
[JRE] That would be a change to RFC 3311 in my opinion. Also the ability to=
 send PAI if it changes (e.g., due to 3PCC transfer) without waiting for th=
e 2xx is, in my opinion, a reasonable current use case. It should be possib=
le to do this without using reliable provisional responses (the only way to=
 get SDP answer, as opposed to just a preview of SDP answer, in a 18x respo=
nse).

John

From shinji.okumura@softfront.jp  Thu May 27 05:31:58 2010
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From: OKUMURA Shinji <shinji.okumura@softfront.jp>
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Date: Thu, 27 May 2010 21:31:47 +0900
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Subject: Re: [sipcore] New deal [ was draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP]
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Christer,

Christer Holmberg <christer.holmberg@ericsson.com>
Thu, 27 May 2010 11:51:38 +0200
>Below are some questions/ideas on how I think we could solve
>most issues. Some of the stuff shall be seen more as brain
>storming than actual proposal at this point.
>
>I am fully aware that some of the ideas for sure may limit
>what is said in RFCs etc, so you don't need to tell me
>about that.
>
>
>
>
>IDEA #1:  Offer/answer: Usage of SDP o= version parameter
>
>I know this has been discussed at some point, but what is
>the reason we can't use the SDP o= version parameter to
>determine in which order the UAS has sent SDPs?
>
>I believe it would solve offer/answer glare situations where
>the UAC does not known in which order the UAS has sent SDPs.
>
>That should not be controversial, and does not break any RFC
>etc.

       A                               B
       |                               |
     s0|re-INV (offer1)                |s0
       |------------------------------>|
       |               18x-rel(answer1)|
       |<------------------------------|
     s1|                        non-2xx|s1
       |                  /------------|<-+
       |                 /  ACK        |s0|
       |                /   ---------->|  |
       |               /    UPD(offer2)|  |
       |<=============/================|  |
       |2xx(answer2) /                 |  |
       |============/=================>|  |
     s2|           /                   |s2|
       |<---------/                    |<-+
     s0|ACK                            |
       |----------->                   |
       |                               |

At least in the above case, o-version seems to be useless for
determining the order. I think the o-version of offer2 should
be the one of answer1 +1.
Even if the UPDATE has the pre-re-INVITE o-version, UA A might
be confused by the decrement.

Regards,
Shinji

>IDEA #2:   Target update: Only outside re-INVITEs
>
>Let's forbid target updates "inside" re-INVITEs. I really
>doubt there is a need for those while the UAs are doing
>something else with the re-INVITE.
>
>That at least solves the glare situation where the remote
>target may be updated in a re-INVITE response and an UPDATE
>response.
>
>
>
>
>IDEA #3:    UPDATE: Always use re-INVITE glare rules
>
>Let's not allow more than one outstanding UPDATE. Ie if a UA
>has sent an UPDATE (with or without SDP), and receives
>another UPDATE request before the answer, it rejects it
>(same rule as for re-INVITE).
>
>
>That would at least solve the problem where a UA sends an
>UPDATE, receives a 200 OK and a UPDATE request, both with
>different target information.
>
>
>Ok, let the tomatoes fly...
>
>...and let me know if there are any issues which the ideas
>above do NOT solve. I have some more ideas, but I am not
>sure they are needed so I'll them to myself for now :)
>
>
>Regards,
>
>Christer

From christer.holmberg@ericsson.com  Thu May 27 09:52:35 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: OKUMURA Shinji <shinji.okumura@softfront.jp>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Thu, 27 May 2010 18:52:22 +0200
Thread-Topic: [sipcore] New deal [ was draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP] - IDEA #1
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Subject: Re: [sipcore] New deal [ was draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP] - IDEA #1
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Hi,

>>IDEA #1:  Offer/answer: Usage of SDP o=3D version parameter
>>
>>I know this has been discussed at some point, but what is
>>the reason we can't use the SDP o=3D version parameter to
>>determine in which order the UAS has sent SDPs?
>>
>>I believe it would solve offer/answer glare situations where
>>the UAC does not known in which order the UAS has sent SDPs.
>>
>>That should not be controversial, and does not break any RFC
>>etc.

       A                                                                  B
       |                                                                   =
 |
     s0|re-INV (offer1)                                           |s0
       |-------------------------------------------->|
       |                                          18x-rel(answer1)|
       |<--------------------------------------------|
     s1|                                                      non-2xx|s1
       |                                                 /------------|<-+
       |                                                 /  ACK        |s0|
       |                                               /   ---------->|    =
|
       |                                            /    UPD(offer2)|    |
       |<=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D/=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D|    |
       |2xx(answer2)                    /                           |    |
       |=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D/=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D>|    |
     s2|                                   /                               =
|s2|
       |<---------------------/                                  |<-+
     s0|ACK                                                           |
       |----------->                                                  |
       |                                                                   =
  |

>At least in the above case, o-version seems to be useless for
>determining the order. I think the o-version of offer2 should
>be the one of answer1 +1.

Yes.

>Even if the UPDATE has the pre-re-INVITE o-version, UA A might
>be confused by the decrement.

I am not sure I understand. The version value shall not be decremented. In =
my understanding the value is only with respect to the previous SDP - you c=
an not use it to switch back to some "old" SDP.

So, even if the UPDATE contains the pre-re-INVITE SDP, the value shall be a=
nswer1 + 1 + 1, because it is a new SDP compared to the previously sent SDP=
 (in the 18x)

Regards,

Christer=

From christer.holmberg@ericsson.com  Thu May 27 10:25:49 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: SIPCORE <sipcore@ietf.org>
Date: Thu, 27 May 2010 19:25:29 +0200
Thread-Topic: Draft new version: draft-ietf-sipcore-keep-03
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Subject: Re: [sipcore] Draft new version: draft-ietf-sipcore-keep-03
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Hi,

Based on the feedback and comments, I intend to produce a new version of th=
e keep draft, with the following changes:

1. Instead of using the Flow-Timer header field, the server recommended kee=
p-alive frequency is inserted as keep parameter value (instead of "yes").

2. If an entity wants to receive keep-alives, it will add the keep paramete=
r value (instead of "yes") in the response, instead of the request. It does=
 not need to be in the request, and it seems to confuse the reader of the d=
raft.

3. The text will say that, when negotiated with REGISTER, the keep-alives a=
re sent until a registration refresh is performed, when the keep-alives nee=
ds to be negotiated again (if the sender still wants to send them). The rea=
son is that it cannot be assumed that a proxy knows whether a REGISTER is f=
or a new registration, or for a registration refresh.

4. The example flows will be updated to reflect the changes above.

5. Editorial changes.

After that, I intend to ask the chairs to initiate WGLC.

Regards,

Christer=

From shinji.okumura@softfront.jp  Thu May 27 18:22:27 2010
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From: OKUMURA Shinji <shinji.okumura@softfront.jp>
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATEwithout SDP
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Gao,

I have a question regarding a target-refresh.

If a UAS return 18x-rel, 2xx to reINVITE and an UPDATE,
and each response has a different Contact-URI(it is a remote
target for a UAC), then how should the UAC handle the UPDATE
and the remote targets?

How does your analyzer work?
Please explain it concretely in order.

    A                               B
    |                               |
    |re-INV (offer1)                |
    |------------------------------>|
    |               18x-rel(answer1)| m:c1
    |<------------------------------|
    |                       2xx(INV)| m:c2
    |<------------------------------|
    |                         UPDATE| m:c3
    |<------------------------------|
    |                               |
or
    A                               B
    |                               |
    |re-INV (offer1)                |
    |------------------------------>|
    |               18x-rel(answer1)| m:c1
    |<------------------------------|
    |                       2xx(INV)| m:c2
    |<---------\ /------------------|
    |           X             UPDATE| m:c3
    |<---------/ \------------------|
    |                               |

Regards,
Shinji

>> How can you only apply "checking for glare situations" during a 
>> glare situation?
>> 
>> Checking for offer/answer glare is part of normal RFC3311 UPDATE 
>processing.
>
>Making two analyzers in series as:
>
>---> analyzer1 -----> analyzer2 ---->
>
>analyzer1 using the rules in RFC3311, and if there's glares case under 
>it's rule, send 491 or (500+retry) directly;
>analyzer2 using the rules in the draft. 
>
>analyzer2 also has two process:
>1. Check whether it can do the ordering of 2xx(INVITE)and UPDATE. If 
>it 
>can do so, pass the UPDATE to the next step normally;
>2. If it can not do the ordering, handle the UPDATE and 2xx(INVITE) as 
>if 
>it contains Offer and using O/A rules to reject.
>
>The effected cases/siutation is only these cases/situation which are 
>ambiguous before the draft.

From shinji.okumura@softfront.jp  Thu May 27 20:39:44 2010
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From: OKUMURA Shinji <shinji.okumura@softfront.jp>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "gao.yang2@zte.com.cn" <gao.yang2@zte.com.cn>, Brett Tate <brett@broadsoft.com>
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Subject: Re: [sipcore] New deal - IDEA [ was draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP]
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Christer,

Christer Holmberg <christer.holmberg@ericsson.com>
Thu, 27 May 2010 11:51:38 +0200
>Below are some questions/ideas on how I think we could solve
>most issues. Some of the stuff shall be seen more as brain
>storming than actual proposal at this point.
>
>I am fully aware that some of the ideas for sure may limit
>what is said in RFCs etc, so you don't need to tell me
>about that.
>
>IDEA #1:  Offer/answer: Usage of SDP o= version parameter
>
>I know this has been discussed at some point, but what is
>the reason we can't use the SDP o= version parameter to
>determine in which order the UAS has sent SDPs?
>
>I believe it would solve offer/answer glare situations where
>the UAC does not known in which order the UAS has sent SDPs.
>
>That should not be controversial, and does not break any RFC
>etc.
>
>IDEA #2:   Target update: Only outside re-INVITEs
>
>Let's forbid target updates "inside" re-INVITEs. I really
>doubt there is a need for those while the UAs are doing
>something else with the re-INVITE.
>
>That at least solves the glare situation where the remote
>target may be updated in a re-INVITE response and an UPDATE
>response.

In the same sequence, since it is not possible to forbid a
offer-anser, the idea seems to be ineffective.
At least it might be difficult to actually use the rule.

For example, when a 2xx response to an UPDATE within an
INVITE transaction include both new remote target and an answer,
how should the UAS handle the response?

>IDEA #3:    UPDATE: Always use re-INVITE glare rules
>
>Let's not allow more than one outstanding UPDATE. Ie if a UA
>has sent an UPDATE (with or without SDP), and receives
>another UPDATE request before the answer, it rejects it
>(same rule as for re-INVITE).

I couldn't agree more.

Regards,
Shinji

>That would at least solve the problem where a UA sends an
>UPDATE, receives a 200 OK and a UPDATE request, both with
>different target information.
>Ok, let the tomatoes fly...
>
>...and let me know if there are any issues which the ideas
>above do NOT solve. I have some more ideas, but I am not
>sure they are needed so I'll them to myself for now :)
>
>Regards,
>
>Christer

From shinji.okumura@softfront.jp  Thu May 27 21:27:00 2010
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From: OKUMURA Shinji <shinji.okumura@softfront.jp>
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Subject: Re: [sipcore] I-D Action:draft-ietf-sipcore-reinvite-04.txt
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Hi all,

Following texts seem to be conflicted.

3.8.  Clarifications on Cancelling Re-INVITEs
   Per
   the rules specified in Section 3.3, if the INVITE request was a re-
   INVITE and some of its requested changes had already been executed,
   the UAS would return a 2xx response instead.

5.5.  UAC Losing its Contact: UAC Behavior
   Such a UAC SHOULD generate a CANCEL request to cancel the re-INVITE
   and cause the INVITE client transaction corresponding to the re-
   INVITE to enter the "Terminated" state.

At least there is a possibility that the conflict occurs.

Regards,
Shinji

Internet-Drafts@ietf.org
Sat,  8 May 2010 00:30:02 -0700 (PDT)
>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-04.txt
>	Pages           : 24
>	Date            : 2010-05-08
>
>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.
>
>A URL for this Internet-Draft is:
>http://www.ietf.org/internet-drafts/draft-ietf-sipcore-reinvite-04.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.

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Subject: Re: [sipcore] I-D Action:draft-ietf-sipcore-reinvite-04.txt
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> Following texts seem to be conflicted.
> 
> 3.8.  Clarifications on Cancelling Re-INVITEs
>    Per
>    the rules specified in Section 3.3, if the INVITE request was a re-
>    INVITE and some of its requested changes had already been executed,
>    the UAS would return a 2xx response instead.

This is about UAS's behavior.

> 
> 5.5.  UAC Losing its Contact: UAC Behavior
>    Such a UAC SHOULD generate a CANCEL request to cancel the re-INVITE
>    and cause the INVITE client transaction corresponding to the re-
>    INVITE to enter the "Terminated" state.

This is about UAC's behavior. And which(2xx/Non-2xx) final response UAS 
would send is defined in section 3.8.

As the UAC may not get the final response from UAS, it should send a new 
Re-INVITE(/UPDATE) to synchronize the state.
And this is defined in the draft.

This is the understanding while I did the review for this part, and I keep 
my understanding as so.


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<br><tt><font size=2><br>
&gt; Following texts seem to be conflicted.<br>
&gt; <br>
&gt; 3.8. &nbsp;Clarifications on Cancelling Re-INVITEs<br>
&gt; &nbsp; &nbsp;Per<br>
&gt; &nbsp; &nbsp;the rules specified in Section 3.3, if the INVITE request
was a re-<br>
&gt; &nbsp; &nbsp;INVITE and some of its requested changes had already
been executed,<br>
&gt; &nbsp; &nbsp;the UAS would return a 2xx response instead.</font></tt>
<br>
<br><tt><font size=2>This is about UAS's behavior.</font></tt>
<br><tt><font size=2><br>
&gt; <br>
&gt; 5.5. &nbsp;UAC Losing its Contact: UAC Behavior<br>
&gt; &nbsp; &nbsp;Such a UAC SHOULD generate a CANCEL request to cancel
the re-INVITE<br>
&gt; &nbsp; &nbsp;and cause the INVITE client transaction corresponding
to the re-<br>
&gt; &nbsp; &nbsp;INVITE to enter the &quot;Terminated&quot; state.</font></tt>
<br>
<br><tt><font size=2>This is about UAC's behavior. And which(2xx/Non-2xx)
final response UAS would send is defined in section 3.8.</font></tt>
<br>
<br><tt><font size=2>As the UAC may not get the final response from UAS,
it should send a new Re-INVITE(/UPDATE) to synchronize the state.</font></tt>
<br><tt><font size=2>And this is defined in the draft.</font></tt>
<br>
<br><tt><font size=2>This is the understanding while I did the review for
this part, and I keep my understanding as so.</font></tt>
<br><br><pre>
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From shinji.okumura@softfront.jp  Thu May 27 23:19:06 2010
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From: OKUMURA Shinji <shinji.okumura@softfront.jp>
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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Gao,

The INVITE transaction has been already completed at the point
to send the UPDATE. In my understanding, the UPDATE can have a
different Contact.

After all, you say that
in 1st chart, UAC may return 2xx to the UPDATE and
in 2nd chart, it should(must?) return 4xx.

Is that right?

Regards,
Shinji

gao.yang2@zte.com.cn
Fri, 28 May 2010 12:45:49 +0800
>Only the first chart is lawful. As the text from RFC3311:
>
>If a UA
>   uses an UPDATE request or response to modify the remote target while
>   an INVITE transaction is in progress, and it is a UAS for that INVITE
>   transaction, it MUST place the same value into the Contact header
>   field of the 2xx to the INVITE that it placed into the UPDATE request
>   or response.
>
>If the UPDATE is in-scope of Re-INVITE, the 2xx's contact should be the 
>same as the (last) UPDATE's contact. So, the second chart can only has the 
>interpretation as the first chart. Then, there's no ambiguous 
>representation here.
>
>OKUMURA Shinji <shinji.okumura@softfront.jp> 
>2010-05-28 09:21
>Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATEwithout SDP
>
>Gao,
>
>I have a question regarding a target-refresh.
>
>If a UAS return 18x-rel, 2xx to reINVITE and an UPDATE,
>and each response has a different Contact-URI(it is a remote
>target for a UAC), then how should the UAC handle the UPDATE
>and the remote targets?
>
>How does your analyzer work?
>Please explain it concretely in order.
>
>    A                               B
>    |                               |
>    |re-INV (offer1)                |
>    |------------------------------>|
>    |               18x-rel(answer1)| m:c1
>    |<------------------------------|
>    |                       2xx(INV)| m:c2
>    |<------------------------------|
>    |                         UPDATE| m:c3
>    |<------------------------------|
>    |                               |
>or
>    A                               B
>    |                               |
>    |re-INV (offer1)                |
>    |------------------------------>|
>    |               18x-rel(answer1)| m:c1
>    |<------------------------------|
>    |                       2xx(INV)| m:c2
>    |<---------\ /------------------|
>    |           X             UPDATE| m:c3
>    |<---------/ \------------------|
>    |                               |
>
>Regards,
>Shinji
>
>>> How can you only apply "checking for glare situations" during a 
>>> glare situation?
>>> 
>>> Checking for offer/answer glare is part of normal RFC3311 UPDATE 
>>processing.
>>
>>Making two analyzers in series as:
>>
>>---> analyzer1 -----> analyzer2 ---->
>>
>>analyzer1 using the rules in RFC3311, and if there's glares case under 
>>it's rule, send 491 or (500+retry) directly;
>>analyzer2 using the rules in the draft. 
>>
>>analyzer2 also has two process:
>>1. Check whether it can do the ordering of 2xx(INVITE)and UPDATE. If 
>>it 
>>can do so, pass the UPDATE to the next step normally;
>>2. If it can not do the ordering, handle the UPDATE and 2xx(INVITE) as 
>>if 
>>it contains Offer and using O/A rules to reject.
>>
>>The effected cases/siutation is only these cases/situation which are 
>>ambiguous before the draft.

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From shinji.okumura@softfront.jp  Fri May 28 00:10:38 2010
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From: OKUMURA Shinji <shinji.okumura@softfront.jp>
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Subject: Re: [sipcore] I-D Action:draft-ietf-sipcore-reinvite-04.txt
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Gao,

I understand.

The text says that the UAC can not expect to receive a 487
response to the INVITE. Right?

And now, assuming that some of changes have already been
executed, the UAS would send a new UPDATE to undo the changes.
But the local target of the UAC has been changed, so I think UAC
should an UPDATE to announce own new target before sending the
CANCEL.

Do you think so?

Regards,
Shinji

gao.yang2@zte.com.cn
Fri, 28 May 2010 13:09:35 +0800
>> Following texts seem to be conflicted.
>> 
>> 3.8.  Clarifications on Cancelling Re-INVITEs
>>    Per
>>    the rules specified in Section 3.3, if the INVITE request was a re-
>>    INVITE and some of its requested changes had already been executed,
>>    the UAS would return a 2xx response instead.
>
>This is about UAS's behavior.
>
>> 
>> 5.5.  UAC Losing its Contact: UAC Behavior
>>    Such a UAC SHOULD generate a CANCEL request to cancel the re-INVITE
>>    and cause the INVITE client transaction corresponding to the re-
>>    INVITE to enter the "Terminated" state.
>
>This is about UAC's behavior. And which(2xx/Non-2xx) final response UAS 
>would send is defined in section 3.8.
>
>As the UAC may not get the final response from UAS, it should send a new 
>Re-INVITE(/UPDATE) to synchronize the state.
>And this is defined in the draft.
>
>This is the understanding while I did the review for this part, and I keep 
>my understanding as so.

From christer.holmberg@ericsson.com  Fri May 28 00:35:00 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: OKUMURA Shinji <shinji.okumura@softfront.jp>, "gao.yang2@zte.com.cn" <gao.yang2@zte.com.cn>, Brett Tate <brett@broadsoft.com>
Date: Fri, 28 May 2010 09:19:16 +0200
Thread-Topic: [sipcore] New deal - IDEA [ was draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP]
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Subject: Re: [sipcore] New deal - IDEA [ was draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP]
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Hi,=20

>>IDEA #2:   Target update: Only outside re-INVITEs
>>
>>Let's forbid target updates "inside" re-INVITEs. I really=20
>>doubt there is a need for those while the UAs are doing something else wi=
th the=20
>>re-INVITE.
>>
>>That at least solves the glare situation where the remote=20
>>target may be updated in a re-INVITE response and an UPDATE response.
>=20
>In the same sequence, since it is not possible to forbid a=20
>offer-anser, the idea seems to be ineffective.
>At least it might be difficult to actually use the rule.
>=20
>For example, when a 2xx response to an UPDATE within an=20
>INVITE transaction include both new remote target and an=20
>answer, how should the UAS handle the response?

As I said, the UPDATE request and/or response within the INVITE transation =
must not include a new remote target.
=20
Regards,

Christer=

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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATEwithout SDP
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Gao,

gao.yang2@zte.com.cn
Fri, 28 May 2010 14:48:19 +0800
>You two charts has difference as:
>
>First chart: 2xx before UPDATE; 
>UAC can treate the UPDATE's contact as the new one.
>
>Second chart: UPDATE before 2xx.
>I guess your puzzle here is about how UAC can order the sequence, that is 
>"UPDATE before 2xx" or "2xx before UPDATE".

I agree this is a puzzle. But your analyzer has to figure out
this puzzle.

>I showed that, if the UPDATE has different target, it MUST be sended after 
>the 2xx by UAS. The reason is that, if UPDATE has a different 
>target(compared as the state before this UPDATE), the 2xx MUST has the 
>same one.(this is defined in RFC3311).
>
>As this, the UAC can treate this UPDATE after 2xx and  UPDATE's target is 
>newer, no matter which one reachs UAC firstly.

After all, you say that in both case UAC may return 2xx to the
UPDATE.

BUT your analyzer2 seems to say "Reject it!".
> 2. If it can not do the ordering, handle the UPDATE and 2xx(INVITE) as 
> if contains Offer and using O/A rules to reject.

Regards,
Shinji

>OKUMURA Shinji <shinji.okumura@softfront.jp> 
>2010-05-28 14:18
>Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
>
>Gao,
>
>The INVITE transaction has been already completed at the point
>to send the UPDATE. In my understanding, the UPDATE can have a
>different Contact.
>
>After all, you say that
>in 1st chart, UAC may return 2xx to the UPDATE and
>in 2nd chart, it should(must?) return 4xx.
>
>Is that right?
>
>Regards,
>Shinji
>
>gao.yang2@zte.com.cn
>Fri, 28 May 2010 12:45:49 +0800
>>Only the first chart is lawful. As the text from RFC3311:
>>
>>If a UA
>>   uses an UPDATE request or response to modify the remote target while
>>   an INVITE transaction is in progress, and it is a UAS for that INVITE
>>   transaction, it MUST place the same value into the Contact header
>>   field of the 2xx to the INVITE that it placed into the UPDATE request
>>   or response.
>>
>>If the UPDATE is in-scope of Re-INVITE, the 2xx's contact should be the 
>>same as the (last) UPDATE's contact. So, the second chart can only has 
>the 
>>interpretation as the first chart. Then, there's no ambiguous 
>>representation here.
>>
>>OKUMURA Shinji <shinji.okumura@softfront.jp> 
>>2010-05-28 09:21
>>Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATEwithout SDP
>>
>>Gao,
>>
>>I have a question regarding a target-refresh.
>>
>>If a UAS return 18x-rel, 2xx to reINVITE and an UPDATE,
>>and each response has a different Contact-URI(it is a remote
>>target for a UAC), then how should the UAC handle the UPDATE
>>and the remote targets?
>>
>>How does your analyzer work?
>>Please explain it concretely in order.
>>
>>    A                               B
>>    |                               |
>>    |re-INV (offer1)                |
>>    |------------------------------>|
>>    |               18x-rel(answer1)| m:c1
>>    |<------------------------------|
>>    |                       2xx(INV)| m:c2
>>    |<------------------------------|
>>    |                         UPDATE| m:c3
>>    |<------------------------------|
>>    |                               |
>>or
>>    A                               B
>>    |                               |
>>    |re-INV (offer1)                |
>>    |------------------------------>|
>>    |               18x-rel(answer1)| m:c1
>>    |<------------------------------|
>>    |                       2xx(INV)| m:c2
>>    |<---------\ /------------------|
>>    |           X             UPDATE| m:c3
>>    |<---------/ \------------------|
>>    |                               |
>>
>>Regards,
>>Shinji
>>
>>>> How can you only apply "checking for glare situations" during a 
>>>> glare situation?
>>>> 
>>>> Checking for offer/answer glare is part of normal RFC3311 UPDATE 
>>>processing.
>>>
>>>Making two analyzers in series as:
>>>
>>>---> analyzer1 -----> analyzer2 ---->
>>>
>>>analyzer1 using the rules in RFC3311, and if there's glares case under 
>>>it's rule, send 491 or (500+retry) directly;
>>>analyzer2 using the rules in the draft. 
>>>
>>>analyzer2 also has two process:
>>>1. Check whether it can do the ordering of 2xx(INVITE)and UPDATE. If 
>>>it 
>>>can do so, pass the UPDATE to the next step normally;
>>>2. If it can not do the ordering, handle the UPDATE and 2xx(INVITE) as 
>>>if 
>>>it contains Offer and using O/A rules to reject.
>>>
>>>The effected cases/siutation is only these cases/situation which are 
>>>ambiguous before the draft.

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From: OKUMURA Shinji <shinji.okumura@softfront.jp>
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Subject: Re: [sipcore] New deal - IDEA [ was draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP]
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Christer,

We have to prepare to recieve something forbidden.
A UAC cannot reject a response to the UPDATE, even if it include
a new remote target. So does the UAC ignore only the target?

Regards,
shinji

Christer Holmberg <christer.holmberg@ericsson.com>
Fri, 28 May 2010 09:19:16 +0200
>Hi, 
>
>>>IDEA #2:   Target update: Only outside re-INVITEs
>>>
>>>Let's forbid target updates "inside" re-INVITEs. I really 
>>>doubt there is a need for those while the UAs are doing something else with the 
>>>re-INVITE.
>>>
>>>That at least solves the glare situation where the remote 
>>>target may be updated in a re-INVITE response and an UPDATE response.
>> 
>>In the same sequence, since it is not possible to forbid a 
>>offer-anser, the idea seems to be ineffective.
>>At least it might be difficult to actually use the rule.
>> 
>>For example, when a 2xx response to an UPDATE within an 
>>INVITE transaction include both new remote target and an 
>>answer, how should the UAS handle the response?
>
>As I said, the UPDATE request and/or response within the
>INVITE transation must not include a new remote target.
> 
>Regards,
>
>Christer

From christer.holmberg@ericsson.com  Fri May 28 00:59:27 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: OKUMURA Shinji <shinji.okumura@softfront.jp>, "sipcore@ietf.org" <sipcore@ietf.org>
Date: Fri, 28 May 2010 09:59:14 +0200
Thread-Topic: [sipcore] New deal - IDEA [ was draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP]
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Subject: Re: [sipcore] New deal - IDEA [ was draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP]
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Hi,=20

>We have to prepare to recieve something forbidden.
>A UAC cannot reject a response to the UPDATE, even if it=20
>include a new remote target. So does the UAC ignore only the target?

Correct - in the same way as it would ignore a "new" SDP answer.

Regards,

Christer


> Christer Holmberg <christer.holmberg@ericsson.com> Fri, 28=20
> May 2010 09:19:16 +0200
> >Hi,
> >
> >>>IDEA #2:   Target update: Only outside re-INVITEs
> >>>
> >>>Let's forbid target updates "inside" re-INVITEs. I really=20
> doubt there=20
> >>>is a need for those while the UAs are doing something else=20
> with the=20
> >>>re-INVITE.
> >>>
> >>>That at least solves the glare situation where the remote=20
> target may=20
> >>>be updated in a re-INVITE response and an UPDATE response.
> >>=20
> >>In the same sequence, since it is not possible to forbid a=20
> >>offer-anser, the idea seems to be ineffective.
> >>At least it might be difficult to actually use the rule.
> >>=20
> >>For example, when a 2xx response to an UPDATE within an INVITE=20
> >>transaction include both new remote target and an answer,=20
> how should=20
> >>the UAS handle the response?
> >
> >As I said, the UPDATE request and/or response within the INVITE=20
> >transation must not include a new remote target.
> >=20
> >Regards,
> >
> >Christer
> =

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> 
> I understand.
> 
> The text says that the UAC can not expect to receive a 487
> response to the INVITE. Right?

As UAC's old contact has been lost, UAC can not expect to receive any 
final response for the old Re-INVITE here.
So, the specific for final response is not important here. And the UAC's 
next Re-INVITE as synchronization can refresh the state here. 

> 
> And now, assuming that some of changes have already been
> executed, the UAS would send a new UPDATE to undo the changes.
> But the local target of the UAC has been changed, so I think UAC
> should an UPDATE to announce own new target before sending the
> CANCEL.
> 
> Do you think so?

I think the UAC should send a target refresh UPDATE before CANCEL. But the 
reason is not about UAS's rollback usage UPDATE.(the state would be 
synchronized by a next Re-INVITE).

To send a new request(the CANCEL) at new address, sending a target refresh 
UPDATE before the new request (the CANCEL) might be better.


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<br><tt><font size=2><br>
&gt; <br>
&gt; I understand.<br>
&gt; <br>
&gt; The text says that the UAC can not expect to receive a 487<br>
&gt; response to the INVITE. Right?</font></tt>
<br>
<br><tt><font size=2>As UAC's old contact has been lost, UAC can not expect
to receive any final response for the old Re-INVITE here.</font></tt>
<br><tt><font size=2>So, the specific for final response is not important
here. And the UAC's next Re-INVITE as synchronization can refresh the state
here. </font></tt>
<br><tt><font size=2><br>
&gt; <br>
&gt; And now, assuming that some of changes have already been<br>
&gt; executed, the UAS would send a new UPDATE to undo the changes.<br>
&gt; But the local target of the UAC has been changed, so I think UAC<br>
&gt; should an UPDATE to announce own new target before sending the<br>
&gt; CANCEL.<br>
&gt; <br>
&gt; Do you think so?</font></tt>
<br>
<br><tt><font size=2>I think the UAC should send a target refresh UPDATE
before CANCEL. But the reason is not about UAS's rollback usage UPDATE.(the
state would be synchronized by a next Re-INVITE).</font></tt>
<br>
<br><tt><font size=2>To send a new request(the CANCEL) at new address,
sending a target refresh UPDATE before the new request (the CANCEL) might
be better.</font></tt>
<br><br><pre>
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> After all, you say that in both case UAC may return 2xx to the
> UPDATE.
> 
> BUT your analyzer2 seems to say "Reject it!".
> > 2. If it can not do the ordering, handle the UPDATE and 2xx(INVITE) as 

> > if contains Offer and using O/A rules to reject.
> 

As the analyzer show is aimed for Brett's puzzle. That is 2xx(of 
Re-INVITE) and UPDATE on different direction, mentioned in section 4.5 of 
the draft. Your case is not so. 

Our aim is to clear up ambiguity case, to cover your case, I'd like to add 
a new restriction highlighted in the analyzer2.

---> analyzer1 -----> analyzer2 ---->

analyzer1 using the rules in RFC3311, and if there's glares case under 
it's rule, send 491 or (500+retry) directly;
analyzer2 using the rules in the draft. 

analyzer2 also has two process:
1. Check whether it can do the ordering of 2xx(INVITE)and UPDATE. If it 
can do so, pass the UPDATE to the next step normally;
2. If there is ambiguity and UAs can not do the ordering, handle the 
UPDATE and 2xx(INVITE) as if it contains Offer and using O/A rules to 
reject.



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<br><tt><font size=2><br>
&gt; After all, you say that in both case UAC may return 2xx to the<br>
&gt; UPDATE.<br>
&gt; <br>
&gt; BUT your analyzer2 seems to say &quot;Reject it!&quot;.<br>
&gt; &gt; 2. If it can not do the ordering, handle the UPDATE and 2xx(INVITE)
as <br>
&gt; &gt; if contains Offer and using O/A rules to reject.<br>
&gt; </font></tt>
<br>
<br><tt><font size=2>As the analyzer show is aimed for Brett's puzzle.
That is 2xx(of Re-INVITE) and UPDATE on different direction, mentioned
in section 4.5 of the draft. Your case is not so. &nbsp;</font></tt>
<br>
<br><tt><font size=2>Our aim is to clear up </font></tt><tt><font size=2 color=blue>ambiguity
case</font></tt><tt><font size=2>, to cover your case, I'd like to add
a new restriction highlighted in the analyzer2.</font></tt>
<br>
<br><tt><font size=2>---&gt; analyzer1 -----&gt; analyzer2 ----&gt;</font></tt>
<br>
<br><tt><font size=2>analyzer1 using the rules in RFC3311, and if there's
glares case under it's rule, send 491 or (500+retry) directly;</font></tt>
<br><tt><font size=2>analyzer2 using the rules in the draft. </font></tt>
<br>
<br><tt><font size=2>analyzer2 also has two process:</font></tt>
<br><tt><font size=2>1. Check whether it can do the ordering of 2xx(INVITE)and
UPDATE. If it can do so, pass the UPDATE to the next step normally;</font></tt>
<br><tt><font size=2>2. If </font></tt><tt><font size=2 color=blue>there
is ambiguity</font></tt><tt><font size=2> and UAs can not do the ordering,
handle the UPDATE and 2xx(INVITE) as if it contains Offer and using O/A
rules to reject.</font></tt>
<br>
<br><br><pre>
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//It seams that mail list doesn't dispatch this mail. To let other people 
catch up the thread, I re-send it.

> After all, you say that in both case UAC may return 2xx to the
> UPDATE.
> 
> BUT your analyzer2 seems to say "Reject it!".
> > 2. If it can not do the ordering, handle the UPDATE and 2xx(INVITE) as 

> > if contains Offer and using O/A rules to reject.
> 

As the analyzer show is aimed for Brett's puzzle. That is 2xx(of 
Re-INVITE) and UPDATE on different direction, mentioned in section 4.5 of 
the draft. Your case is not so. 

Our aim is to clear up ambiguity case, to cover your case, I'd like to add 
a new restriction highlighted in the analyzer2.

---> analyzer1 -----> analyzer2 ---->

analyzer1 using the rules in RFC3311, and if there's glares case under 
it's rule, send 491 or (500+retry) directly;
analyzer2 using the rules in the draft. 

analyzer2 also has two process:
1. Check whether it can do the ordering of 2xx(INVITE)and UPDATE. If it 
can do so, pass the UPDATE to the next step normally;
2. If there is ambiguity and UAs can not do the ordering, handle the 
UPDATE and 2xx(INVITE) as if it contains Offer and using O/A rules to 
reject.



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<br><tt><font size=2>//It seams that mail list doesn't dispatch this mail.
To let other people catch up the thread, I re-send it.</font></tt>
<br><tt><font size=2><br>
&gt; After all, you say that in both case UAC may return 2xx to the<br>
&gt; UPDATE.<br>
&gt; <br>
&gt; BUT your analyzer2 seems to say &quot;Reject it!&quot;.<br>
&gt; &gt; 2. If it can not do the ordering, handle the UPDATE and 2xx(INVITE)
as <br>
&gt; &gt; if contains Offer and using O/A rules to reject.<br>
&gt; </font></tt>
<br>
<br><tt><font size=2>As the analyzer show is aimed for Brett's puzzle.
That is 2xx(of Re-INVITE) and UPDATE on different direction, mentioned
in section 4.5 of the draft. Your case is not so. &nbsp;</font></tt>
<br>
<br><tt><font size=2>Our aim is to clear up </font></tt><tt><font size=2 color=blue>ambiguity</font></tt><tt><font size=2>
case, to cover your case, I'd like to add a new restriction highlighted
in the analyzer2.</font></tt>
<br>
<br><tt><font size=2>---&gt; analyzer1 -----&gt; analyzer2 ----&gt;</font></tt>
<br>
<br><tt><font size=2>analyzer1 using the rules in RFC3311, and if there's
glares case under it's rule, send 491 or (500+retry) directly;</font></tt>
<br><tt><font size=2>analyzer2 using the rules in the draft. </font></tt>
<br>
<br><tt><font size=2>analyzer2 also has two process:</font></tt>
<br><tt><font size=2>1. Check whether it can do the ordering of 2xx(INVITE)and
UPDATE. If it can do so, pass the UPDATE to the next step normally;</font></tt>
<br><tt><font size=2>2. If </font></tt><tt><font size=2 color=blue>there
is ambiguity</font></tt><tt><font size=2> and UAs can not do the ordering,
handle the UPDATE and 2xx(INVITE) as if it contains Offer and using O/A
rules to reject.</font></tt>
<br>
<br><br><pre>
--------------------------------------------------------
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//It seams that mail list doesn't dispatch this mail. To let other people 
catch up the thread, I re-send it.


> 
> I understand.
> 
> The text says that the UAC can not expect to receive a 487
> response to the INVITE. Right?

As UAC's old contact has been lost, UAC can not expect to receive any 
final response for the old Re-INVITE here.
So, the specific for final response is not important here. And the UAC's 
next Re-INVITE as synchronization can refresh the state here. 

> 
> And now, assuming that some of changes have already been
> executed, the UAS would send a new UPDATE to undo the changes.
> But the local target of the UAC has been changed, so I think UAC
> should an UPDATE to announce own new target before sending the
> CANCEL.
> 
> Do you think so?

I think the UAC should send a target refresh UPDATE before CANCEL. But the 
reason is not about UAS's rollback usage UPDATE.(the state would be 
synchronized by a next Re-INVITE).

To send a new request(the CANCEL) at new address, sending a target refresh 
UPDATE before the new request (the CANCEL) might be better.


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<br><tt><font size=2>//It seams that mail list doesn't dispatch this mail.
To let other people catch up the thread, I re-send it.</font></tt>
<br>
<br><tt><font size=2><br>
&gt; <br>
&gt; I understand.<br>
&gt; <br>
&gt; The text says that the UAC can not expect to receive a 487<br>
&gt; response to the INVITE. Right?</font></tt>
<br>
<br><tt><font size=2>As UAC's old contact has been lost, UAC can not expect
to receive any final response for the old Re-INVITE here.</font></tt>
<br><tt><font size=2>So, the specific for final response is not important
here. And the UAC's next Re-INVITE as synchronization can refresh the state
here. </font></tt>
<br><tt><font size=2><br>
&gt; <br>
&gt; And now, assuming that some of changes have already been<br>
&gt; executed, the UAS would send a new UPDATE to undo the changes.<br>
&gt; But the local target of the UAC has been changed, so I think UAC<br>
&gt; should an UPDATE to announce own new target before sending the<br>
&gt; CANCEL.<br>
&gt; <br>
&gt; Do you think so?</font></tt>
<br>
<br><tt><font size=2>I think the UAC should send a target refresh UPDATE
before CANCEL. But the reason is not about UAS's rollback usage UPDATE.(the
state would be synchronized by a next Re-INVITE).</font></tt>
<br>
<br><tt><font size=2>To send a new request(the CANCEL) at new address,
sending a target refresh UPDATE before the new request (the CANCEL) might
be better.</font></tt>
<br><br><pre>
--------------------------------------------------------
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Cc: "Christer.Holmberg@ericsson.com" <Christer.Holmberg@ericsson.com>, "sipcore@ietf.org" <sipcore@ietf.org>
Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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Sorry to be so slow in responding. I got behind, and had to wait till I 
had time to read through all that has been said before I could make any 
sort of meaningful response.

As it is, there is so much that I am responding after reading most of 
the -reinvite- traffic, without first checking if some of the mail with 
different subject is also relevant.

gao.yang2@zte.com.cn wrote:
> 
> Rules for "When checking for glare situations" in the draft text means 
> the UAs's reasoning and handling while the UAs are in a glare 
> situations, which is ambiguous.
> 
> It does not means UAs reasoning and handling for every use case to 
> figure out whether this case is a glare case.
> 
> This is my understanding. And I'd like other co-authors and important 
> reviewer to comment on this point.

In light of all the discussion this has generated, I think -reinvite- is 
not sufficiently clear on what it means. I think we need to be looking 
for a way to revise it to be more clear, regardless of its current 
post-WGLC status. (Note that -offeranswer- is also in post-WGLC status, 
and we are debating changes to it.)

We *REALLY* need to hear from Gonzalo for his opinion on this!

My opinion at this point is that stating that "UPDATEs without SDP 
should be treated as if they contained an offer" is an insufficient 
technique for stating the required behavior - that it needs to be stated 
more explicitly, even though that will take more words.

I *think* that considerations of *glare* occur when there is something 
to be *negotiated* and either side may initiate the negotiation. The 
negotiated piece of state that has been considered to date for glare is 
the pair of SDP bodies that describe the media sessions.

As noted elsewhere in this thread, another piece state that is 
*negotiated* is session-timer. As a result, there could be a glare 
situation around that. The determination of whether glare has occurred 
ought to be independent for each kind of negotiated state. So one 
determination for SDP, another for session-timer. But there is only one 
mechanism for signaling the glare in either case, the 491 response code. 
So it would be used if *either* sdp-glare or session-timer-glare occurred.

Target-refresh is somewhat different, because it isn't *negotiated* - it 
is simply *declared* independently in each direction. So I don't think 
it really ought to be considered a glare situation.

Another in this category is sending an updated PAI, or changing From. 
This to is a declaration rather than a negotiation, so that glare 
shouldn't apply.

In both of those cases, there remains an ordering issue, and perhaps the 
answer is to forbid behavior that can result in ambiguity. But I fear 
that cure is worse than the disease, so I'm inclined to seek a better cure.

I'm especially concerned about that in the case of target refresh. If 
the sender has lost its old address, then we are on the brink of losing 
the call entirely. Saving the call is more important than whether we 
have the proper o/a state - that can be fixed after the new target 
address is established. Also, target refresh cases are typically rare. 
Two successive ones in close succession in the same direction should be 
rare^2. (A target refresh of both ends in quick succession could be more 
likely - for instance if the same event causes both to lose their 
address. It seems unlikely that this will be a recoverable event.)

So, I think we ought not be forbidding target refresh just because of 
other state. Ambiguity about which is the old target and which is the 
new one can be measured by which is a change - assuming there aren't two 
changes in quick succession. I think Gao has also recommended this.

	Thanks,
	Paul



>  > > > Look at RFC3311 (especially section 5.2) concerning how to behave
>  > > > when UPDATE offer is received while there is an outstanding offer.
>  > >
>  > > I know this part of 3311. I don't know the meaning of why
>  > > you highlight this part.
>  > >
>  > > Do you think text of our draft has conflict with this part of 3311 
> here?
>  > >
>  > > I don't think so, as our text only for ambiguous cases.
>  >
>  > I'm indicating that the RFC3311 mandates the returning 491 or 500
>  > (depending upon direction).
>  >
>  > You are indicating that the RFC3311 text doesn't apply even though
>  > draft-ietf-sipcore-reinvite-04 sections 3.5 and 4.5 indicate they
>  > would apply.  RFC3311 section 5.2 includes glare checking as part of
>  > normal UPDATE processing.
>  >
>  > The drafts normative text causing the issue: "When checking for
>  > glare situations, UAs MUST treat every UPDATE request as if it
>  > contained an offer."
>  >
>  > You keep interpreting/indicating that the normative text does not
>  > apply to all UPDATEs.  Please provide the normative text which
>  > indicates your interpretation.
>  >
>  > Thanks,
>  > Brett
>  >
>  >
> 
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From pkyzivat@cisco.com  Sun May 30 14:34:41 2010
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Cc: sipcore@ietf.org, Harbhanu <harbhanu@huawei.com>
Subject: Re: [sipcore] REg. draft-ietf-sipcore-reinvite-04  //Session Timer
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I agree that there are issues with session-timer.

Normally session-timer is simple, because it is negotiated anew with
each exchange. That works reasonably well for reINVITE and for UPDATE
outside of an invite transaction. Its a mess if an UPDATE within a
(re)INVITE transaction negotiates S-T. I think that needs to be
forbidden, which will be a revision to 4028.

(A reINVITE should not take so long that a session-timer will expire
before it is complete. Perhaps it should be specified that *if* a
session-timer does expire while a reINVITE is in progress, that
expiration is ignored.)

	Thanks,
	Paul

gao.yang2@zte.com.cn wrote:
> 
> Hi,
> 
> Not like O/A usage and target refresh usage, there's no explicit 
> definition of allowing subsequent session refresh request in embedded 
> requests(such as UPDATE during pending Re-INVITE). So, I think it is not 
> recommended to sending UPDATE as subsequent session refresh request in 
> pending INVITE/Re-INVITE.
> 
> I'd like to talk more about session timer. If people want to do the 
> semantics extension for session timer's to allow embedded requests(such 
> as UPDATE during pending Re-INVITE), there's one suggestion 
> way(INFORMATIVE at all):
> 
> Do it like target refresh, then the 2xx of INVITE/Re-INVITE's SE value 
> should be as the SE value in the last embedded subsequent session 
> refresh request's 2xx.
> 
> Then, the glare handling for such usage would be the same as the Target 
> Refresh part in draft-ietf-sipcore-reinvite.
> 
> Thanks,
> 
> Gao
> 
> ===================================
> Zip    : 210012
> Tel    : 87211
> Tel2   :(+86)-025-52877211
> e_mail : gao.yang2@zte.com.cn
> ===================================
> 
> 
> *Harbhanu <harbhanu@huawei.com>*
> ·¢¼þÈË:  sipcore-bounces@ietf.org
> 
> 2010-05-24 13:40
> 
> 	
> ÊÕ¼þÈË
> 	sipcore@ietf.org
> ³­ËÍ
> 	
> Ö÷Ìâ
> 	[sipcore] REg. draft-ietf-sipcore-reinvite-04
> 
> 
> 	
> 
> 
> 
> 
> 
>  
> 1. UE1 --- INVITE SE-1800 ¨¤ UE2
> 2. UE1 <--- 183 ------------------ UE2
>  
> 3. UE1 --- UPDATE SE-600 ¨¤ UE2
> 4. UE1 <--- 200 UPDATE -SE600 -- UE2
>  
> 5. UE1 <--- 200 INVITE SE-1800 -- UE2
>  
> Please refer the scenario above for handling of Session-Timer, (4028).
> As per the Draft-ietf-sipcore-reinvite-04 section 4.5:
> ¡°  When checking for glare situations, UAs MUST treat every UPDATE
>    request as if it contained an offer.  Additionally, UAs MUST treat
>    the exchange of a 2xx response to an INVITE request and its
>    corresponding ACK request as an offer/answer exchange.  Consequently,
>    the rules regarding glare situations applicable to offer/answer
>    exchanges are also applicable to those exchanges.¡±
>  
> Here, we are not able to conclude whether both UAs should run session 
> timer with
> 1800 or 600, or otherwise whether the UPDATE should be rejected.
>  
> As per my understanding this draft doesn¡¯t **explicitly** mention 
> anything about the Session-Timer, wherein there is an early negotiation. 
> Since 4028 also doesn¡¯t
> have any reference for early negotiation, so we suggest that it can 
> better be incorporated in the above mentioned 
> draft[ietf-sipcore-reinvite-04].
>  
> Please share your opinion and do guide us incase any of the existing 
> RFC/active-draft already captures this.
>  
> Thanks!
>  
> Regards,
> Harbhanu
>  
> ***************************************************************************************
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From pkyzivat@cisco.com  Sun May 30 14:51:21 2010
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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This is evidence for my assertion that glare is really about conflicts 
in negotiation of state, and not about the specific messages used to 
conduct that negotiation.

	Thanks,
	Paul

Brett Tate wrote:
>> Sorry, I can't catch the meaning of "when you factor 
>> in RFC3311". Could you show me the meaning?
> 
> Look at RFC3311 (especially section 5.2) concerning how to behave when UPDATE offer is received while there is an outstanding offer.
> 
> " If an UPDATE is received that contains an offer, and the UAS has
>  generated an offer (in an UPDATE, PRACK or INVITE) to which it has
>  not yet received an answer, the UAS MUST reject the UPDATE with a 491
>  response.  Similarly, if an UPDATE is received that contains an
>  offer, and the UAS has received an offer (in an UPDATE, PRACK, or
>  INVITE) to which it has not yet generated an answer, the UAS MUST
>  reject the UPDATE with a 500 response, and MUST include a Retry-After
>  header field with a randomly chosen value between 0 and 10 seconds."
> 
> 
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From pkyzivat@cisco.com  Sun May 30 15:13:01 2010
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Subject: Re: [sipcore] New deal [ was draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP]
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Christer Holmberg wrote:
> Below are some questions/ideas on how I think we could solve most issues. Some of the stuff shall be seen more as brain storming than actual proposal at this point.
> 
> I am fully aware that some of the ideas for sure may limit what is said in RFCs etc, so you don't need to tell me about that.
> 
> 
> 
> 
> IDEA #1:  Offer/answer: Usage of SDP o= version parameter
> 
> I know this has been discussed at some point, but what is the reason we can't use the SDP o= version parameter to determine in which order the UAS has sent SDPs?
> 
> I believe it would solve offer/answer glare situations where the UAC does not known in which order the UAS has sent SDPs.
> 
> That should not be controversial, and does not break any RFC etc.

Shinji refuted this one.

> IDEA #2:   Target update: Only outside re-INVITEs
> 
> Let's forbid target updates "inside" re-INVITEs. I really doubt there is a need for those while the UAs are doing something else with the re-INVITE.
> 
> That at least solves the glare situation where the remote target may be updated in a re-INVITE response and an UPDATE response.

For the case of a UA that loses its IP address and must switch to 
another, it may be essential to send the UPDATE immediately. And it 
won't be getting the response to the reINVITE in that case, so it can't 
wait till it does.

In general you should almost never ignore a target refresh, unless there 
is something grievous such as an authentication failure.

> IDEA #3:    UPDATE: Always use re-INVITE glare rules
> 
> Let's not allow more than one outstanding UPDATE. Ie if a UA has sent an UPDATE (with or without SDP), and receives another UPDATE request before the answer, it rejects it (same rule as for re-INVITE).
> 
> 
> That would at least solve the problem where a UA sends an UPDATE, receives a 200 OK and a UPDATE request, both with different target information.

I'm not convinced this solves anything. And if it prevented an essential 
target refresh then it would be a bad thing.

As I stated elsewhere, I think glare should be detected individually for 
each piece of state being negotiated. Then the request can be rejected 
if some form of glare has been detected.

Target refresh isn't a glare situation, so would never fall into that.

	Thanks,
	Paul

> Ok, let the tomatoes fly...
> 
> ...and let me know if there are any issues which the ideas above do NOT solve. I have some more ideas, but I am not sure they are needed so I'll them to myself for now :)
> 
> 
> Regards,
> 
> Christer
> 
> 
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 

From pkyzivat@cisco.com  Sun May 30 15:16:45 2010
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE	without SDP
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I read through this discussion, and I cannot figure out what the 
difference is between the two of you. As of this reply it seems to me 
that the two of you are agreeing.

Can one of you restate the issue and the contrasting solutions?

	Thanks,
	Paul

OKUMURA Shinji wrote:
> Gao,
> 
> This is a figure I've assumed.
> 
>     A                               B
>     |                               |
>   s0|re-INV (offer1)                |s0
>     |------------------------------>|
>     |               18x-rel(answer1)|
>     |<------------------------------|
>   s1|                        non-2xx|s1
>     |<------------------------------|<-+
>   s0|ACK                            |s0|
>     |----------->                   |  |
>     |UPD(offer0)                    |  |
>     |==============================>|* | reject UPDATE
>     |                       491(UPD)|  |
>     |<==============================|  |
>     |                    ACK        |  |
>     |                    ---------->|<-+
>     |                               |
>                 Figure 2
> 
>> Hi,
>>
>> OKUMURA Shinji <shinji.okumura@softfront.jp>
>>> Gao,
>>>
>>> gao.yang2@zte.com.cn
>>> Tue, 25 May 2010 14:15:10 +0800
>>>> Hi,
>>>>
>>>>>>> I agree that it doesn't forbid sending it.  It allows it and 
>>>>>>> mandates the UAS return 491 or 500 (depending upon direction).
>>>>>> It mandates the UAS send 4xx/5xx to UPDATE only when the UAS
>>>>>> send 4xx/5xx to the Re-INVITE and the UAS get the UPDATE before
>>>>>> ACK of 4xx/5xx(Re-INVITE).
>>>>> As you know, ACK to non-2xx is hop-by-hop.
>>>>> There is little benefit to reject such an UPDATE.
>>>> UAS send 491 for the UPDATE(in the flow) to *tell* UAC that in it's
>>>> (UAS) 
>>>> view, there's a glare case.
>>> But this UPDATE is sent in order to synchronize the common view.
>>> If UAS return 2xx, then the situation goes into nonglare.
>> I guess I have a inverse point here. By this draft's clarification, 
>> there's no inconsistent session state for the non-successful Re-INVITE. 
>> So, why UAC send a UPDATE to synchronize the common view?
> 
> In 3.4.  UAC Behavior,
>   "In order to cope with these race condition
>    situations, a UAC that receives an error response to a re-INVITE for
>    which changes have been already executed SHOULD generate a new re-
>    INVITE or UPDATE request in order to make sure that both UAs have a
>    common view of the state of the dialog and the session."
> 
> Under the above descriptions, this UAC generates a new UPDATE.
> 
>> While if UAS return 2xx(for the UPDATE), the situation would get into 
>> glare(UAS has no way to figure out this UPDATE is in-scope of Re-INVITE or 
>> not). This is relating to atomic of session state. I'd like to emphasize 
>> that keeping Re-INVITE atomic, except for target refresh, is the main idea 
>> in this draft.
> 
> This issue has nothing to do with the "atomic of session state".
> 
>>> Probably another UPDATE will be sent, but this UPDATE
>>> transaction is supposed to be avoided.
>>>
>>> At the very start, the UAS rejects a reINVITE does not care
>>> about this text. Is it necessary to let such a UAS follow a
>>> new rule?
>> These days, you guys showed a lot of use cases. I don't remember the 
>> details. Could you say more about this use case.
> 
> Please look at the original mail of mine, and re-comment it.
> You seem to be confused a little.
> 
>>>> When UAS get the ACK then UPDATE, it still has no precise criterion for 
>>>> whether the UAC gets non-2xx(Re-INVITE) first then send the UPDATE or not. 
>>>> Keeping rejecting the UPDATE is not a reasonable way here. So, the draft 
>>>> suggest the way, as UAS can send 2xx to the UPDATE.
>>> It is a conventional handling and enough for UAS.
>>>
>>>> When UAC get UAS's 4xx of UPDATE, the session state is clear for it. And 
>>>> UAC would only send a synchronizing Re-INVITE/UPDATE when it is a glare 
>>>> case in the UAC's view.
>>> UAS' behavior does not affect UAC' behavior directly.
>>> UAC will do the needful regardless of whether UAS rejects
>>> a reINVITE.
>>>
>>>> IMO, this way is better than the way we might once use: allowing B send
>>>> 2xx to UPDATE, and both A and B *see* glare, and all send Re-INVITE/UPDATE 
>>>> to refresh session state by each's view.
>>> The new rule for UAS does not seem to improve the situation.
>>> The new rule for UAC seems to be just enough to synchronize
>>> the view.
>> The key merit for the new rule is make both UAC and UAS clear in which 
>> situation is it's responsibility to handle glare case, and there's only 
>> one UA need to do something. Both sides want to refresh session state by 
>> each's view is not a elegant way.
> 
> The crux of the matter is that a UAS can not recognize own glare
> situation anytime.
> It is UAC's duty to synchronize a common view.
> 
> Regards,
> Shinji
> 
>> Thanks,
>>
>> Gao
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 

From pkyzivat@cisco.com  Sun May 30 15:28:43 2010
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE	withoutSDP
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Shinji,

I agree with what you say, but perhaps for different reasons, for based 
on different view of what "glare" means.

IMO the 491 response should be used to indicate glare has been detected, 
specifically that a proposal to negotiate a state change has been 
*received* while a previously *sent* proposal to negotiate a state 
change is still unresolved. That can apply to o/a regardless of what 
messages cause the situation to be detected. And I think this can apply 
even when receiving an offerless reINVITE, which is a demand for an offer.

OTOH the 500 response applies when a proposal to negotiate a state 
change has been *received* while a previously *received* proposal to 
negotiate state is still unresolved. This is not glare, because the 
problem is not the result of coincidence in valid behavior. The 500 is a 
suitable response because it is bad behavior by the sender of those 
proposals that results in the problem.

	Thanks,
	Paul

OKUMURA Shinji wrote:
> Hi Brett,
> 
> I have a similar question too.
> 
> RFC 3261 and 3311 describe the definition regarding 500 and
> 491 response.
> When UA rejects the INVITE during an INVITE transaction, or
> rejects the UPDATE during an UPDATE transaction, it has nothing
> to do with offer-answer which error code to be used.
> 
> If both transaction have the same direction, the error code is 500.
> If not, the error code is 491.
> 
> But an interworking of INVITE and UPDATE is unclear.
> The draft-offeranswer makes an efforts to clarify it in the
> section 4.3.
> 
>> Howdy,
>>
>> Draft-ietf-sipcore-reinvite-04 sections 3.5 and 4.5 indicate
>> that an UPDATE without SDP should be treated (from a glare
>> perspective) as though it contained an offer SDP.
>> However I currently find the definition of "glare" and the
>> expected behavior to be ambiguous.
>>
>> What is the precise meaning of glare?  I ask mainly because
>> some vendors send a 491 when an RFC indicates to send a 500.
>>
>> If INVITE contains offer SDP and 18x was not reliable, how
>> SHOULD the called party respond if UPDATE without SDP is
>> received from caller: 491, 500, or process as though no glare?
> 
> If UA must refuse it , parhaps it is 500.
> IMO it is another question whether UA must reject the UPDATE
> without an offer.
> 
>> If INVITE contains offer SDP and 18x was not reliable, how
>> SHOULD the caller respond if UPDATE without SDP is received
>>from called party: 491, 500, or process as though no glare?
> 
> Parhaps it is 491.
> 
>> If caller receives INVITE 2xx without offer/answer, how SHOULD
>> or MUST the caller respond if UPDATE without SDP is received
>>from called party prior to sending ACK (if not willing to
>> queue or drop UPDATE): 491 or 500?  Is it a SHOULD or MUST?
> 
> UA may send 2xx to UPDATE. Because the INVITE transaction has
> been completed.
> 
> For reference, the following figures are now under consideration.
> 
>   A                      B
>   |                      |
>   |INVITE                |
>   |--------------------->|<-+             <-----+
>   |  1xx(offer or answer)|  |                   |
>   |<---------------------|  | reject INVITE     | reject UPDATE
>   |PRACK                 |  | Don't send INVITE | Don't send UPDATE
>   |--------------------->|  |                   |
>   |               2xx-PRA|  |                   |
>   |<---------------------|  |             <-----+
>   |                      |  |                   | may send UPDATE
>   |                   2xx|  |                   | may accept UPDATE
>   |<---------------------|<-+                   |
>   |                      |  | may send INVITE   |
>   |ACK                   |  | may accept INVITE |
>   |--------------------->|  |                   |
>   |                      |  v                   v
>   |                      |
>           Figure 16
> 
>   A                      B
>   |                      |
>   |INVITE                |
>   |--------------------->|<-+             <-----+
>   |  1xx(offer or answer)|  |                   |
>   |<---------------------|  | reject INVITE     | reject UPDATE
>   |PRACK                 |  | Don't send INVITE | Don't send UPDATE
>   |--------------------->|  |                   |
>   |               2xx-PRA|  |                   |
>   |<---------------------|  |             <-----+
>   |                      |  |                   | may send UPDATE
>   |                   4xx|  |                   | may acceptUPDATE
>   |<---------------------|<-+                   |
>   |                      |  | Don't send INVITE |
>   |ACK          ACK      |  | may accept INVITE | <-- Should B reject UPDATE or not?
>   |--------->   -------->|<-+             <-----+     That is another question.
>   |                      |  |                   |
>   |                      |  v                   v
>           Figure 17
> 
> In both figures, 
> UA A reject all INVITE/UPDATE with 491.
> UA B reject all INVITE/UPDATE with 500.
> 
> Regards,
> Shinji
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 

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Subject: Re: [sipcore] I-D Action:draft-ietf-sipcore-reinvite-04.txt
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I think its important to send the target refresh ASAP (if the old target 
is no longer functional.)

Whether sending it before/after CANCEL otherwise makes little 
difference. Either way the 487 won't be received.

	Thanks,
	Paul

OKUMURA Shinji wrote:
> Gao,
> 
> I understand.
> 
> The text says that the UAC can not expect to receive a 487
> response to the INVITE. Right?
> 
> And now, assuming that some of changes have already been
> executed, the UAS would send a new UPDATE to undo the changes.
> But the local target of the UAC has been changed, so I think UAC
> should an UPDATE to announce own new target before sending the
> CANCEL.
> 
> Do you think so?
> 
> Regards,
> Shinji
> 
> gao.yang2@zte.com.cn
> Fri, 28 May 2010 13:09:35 +0800
>>> Following texts seem to be conflicted.
>>>
>>> 3.8.  Clarifications on Cancelling Re-INVITEs
>>>    Per
>>>    the rules specified in Section 3.3, if the INVITE request was a re-
>>>    INVITE and some of its requested changes had already been executed,
>>>    the UAS would return a 2xx response instead.
>> This is about UAS's behavior.
>>
>>> 5.5.  UAC Losing its Contact: UAC Behavior
>>>    Such a UAC SHOULD generate a CANCEL request to cancel the re-INVITE
>>>    and cause the INVITE client transaction corresponding to the re-
>>>    INVITE to enter the "Terminated" state.
>> This is about UAC's behavior. And which(2xx/Non-2xx) final response UAS 
>> would send is defined in section 3.8.
>>
>> As the UAC may not get the final response from UAS, it should send a new 
>> Re-INVITE(/UPDATE) to synchronize the state.
>> And this is defined in the draft.
>>
>> This is the understanding while I did the review for this part, and I keep 
>> my understanding as so.
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 

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From: Paul Kyzivat <pkyzivat@cisco.com>
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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Elwell, John wrote:
> If an UPDATE request without SDP offer cannot be accepted when only an unreliable provisional response has occurred, I believe this is a normative change to RFC 3311.

Yes, I agree. And I think its an undesirable change too.

	Thanks,
	Paul

> John
> 
>> -----Original Message-----
>> From: Brett Tate [mailto:brett@broadsoft.com] 
>> Sent: 24 May 2010 14:25
>> To: gao.yang2@zte.com.cn
>> Cc: Elwell, John; sipcore@ietf.org
>> Subject: RE: [sipcore] draft-ietf-sipcore-reinvite-04: glare 
>> and UPDATE without SDP
>>
>>>> Concerning glare, my understanding is that the draft mandates 
>>>> treating UPDATE without offer as though it contained an offer.
>>>>
>>>> I'm still trying to understand how section 3.5 and 4.5 are to be 
>>>> interpreted.  Because the draft uses offer/answer rules for the 
>>>> "fix", it prevents using UPDATE without SDP on an early 
>> dialog until
>>>> allowed per offer/answer rules. 
>>> Not exactly here. The text never forbit usage of UPDATE 
>> without offer by O/A rules. 
>>
>> I agree that it doesn't forbid sending it.  It allows it and 
>> mandates the UAS return 491 or 500 (depending upon direction).
>>
>>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore
> 

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From pkyzivat@cisco.com  Sun May 30 18:09:16 2010
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Date: Sun, 30 May 2010 21:09:01 -0400
From: Paul Kyzivat <pkyzivat@cisco.com>
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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gao.yang2@zte.com.cn wrote:
> 
> I guess you are talking about section 4.5 of the draft.

Yes.

> But section 4.5's rules only apply to case which has ambiguity. I think 
> that the talked use case here has no ambiguity. So, nothing is forbidden 
> here.

I don't know for certain when those rules apply. It says "When checking
for glare conditions". That is too vague for me. Under what
circumstances does the UA check for glare conditions?

My longer message on glare, etc. is probably a better context within
which to have this discussion.

	Thanks,
	Paul

> Thanks,
> 
> Gao
> 
> ===================================
> Zip    : 210012
> Tel    : 87211
> Tel2   :(+86)-025-52877211
> e_mail : gao.yang2@zte.com.cn
> ===================================
> 
> 
> *Paul Kyzivat <pkyzivat@cisco.com>*
> ·¢¼þÈË:  sipcore-bounces@ietf.org
> 
> 2010-05-31 08:03
> 
> 	
> ÊÕ¼þÈË
> 	"Elwell, John" <john.elwell@siemens-enterprise.com>
> ³­ËÍ
> 	"gao.yang2@zte.com.cn" <gao.yang2@zte.com.cn>, "sipcore@ietf.org" 
> <sipcore@ietf.org>
> Ö÷Ìâ
> 	Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
> 
> 
> 	
> 
> 
> 
> 
> 
> 
> 
> Elwell, John wrote:
>  > If an UPDATE request without SDP offer cannot be accepted when only 
> an unreliable provisional response has occurred, I believe this is a 
> normative change to RFC 3311.
> 
> Yes, I agree. And I think its an undesirable change too.
> 
>                 Thanks,
>                 Paul
> 
>  > John
>  >
>  >> -----Original Message-----
>  >> From: Brett Tate [mailto:brett@broadsoft.com]
>  >> Sent: 24 May 2010 14:25
>  >> To: gao.yang2@zte.com.cn
>  >> Cc: Elwell, John; sipcore@ietf.org
>  >> Subject: RE: [sipcore] draft-ietf-sipcore-reinvite-04: glare
>  >> and UPDATE without SDP
>  >>
>  >>>> Concerning glare, my understanding is that the draft mandates
>  >>>> treating UPDATE without offer as though it contained an offer.
>  >>>>
>  >>>> I'm still trying to understand how section 3.5 and 4.5 are to be
>  >>>> interpreted.  Because the draft uses offer/answer rules for the
>  >>>> "fix", it prevents using UPDATE without SDP on an early
>  >> dialog until
>  >>>> allowed per offer/answer rules.
>  >>> Not exactly here. The text never forbit usage of UPDATE
>  >> without offer by O/A rules.
>  >>
>  >> I agree that it doesn't forbid sending it.  It allows it and
>  >> mandates the UAS return 491 or 500 (depending upon direction).
>  >>
>  >>
>  > _______________________________________________
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>  > sipcore@ietf.org
>  > https://www.ietf.org/mailman/listinfo/sipcore
>  >
> _______________________________________________
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> 
> 
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> > I guess you are talking about section 4.5 of the draft.
> 
> Yes.
> 
> > But section 4.5's rules only apply to case which has ambiguity. I 
think 
> > that the talked use case here has no ambiguity. So, nothing is 
forbidden 
> > here.
> 
> I don't know for certain when those rules apply. It says "When checking
> for glare conditions". That is too vague for me. Under what
> circumstances does the UA check for glare conditions?

Yes. I also think the words is vague. 
So, I think we may need *words enhancement*, if my understanding of it's 
technical semantics is correct.
If it's technical semantics is not correct or not comprehensive enough, we 
might need *words and technical enhancement* at the same time here.

My point here is that as ther's arguement about it's technical semantics's 
correctness(IMO, section 3.5 and 4.5 is correct, technically), it would be 
more acceptable for me to describe it as "its an *potential* undesirable 
change too".

> 
> My longer message on glare, etc. is probably a better context within
> which to have this discussion.

I am reading it, and will go on soon.

Thanks,

Gao


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<br><tt><font size=2><br>
&gt; &gt; I guess you are talking about section 4.5 of the draft.<br>
&gt; <br>
&gt; Yes.<br>
&gt; <br>
&gt; &gt; But section 4.5's rules only apply to case which has ambiguity.
I think <br>
&gt; &gt; that the talked use case here has no ambiguity. So, nothing is
forbidden <br>
&gt; &gt; here.<br>
&gt; <br>
&gt; I don't know for certain when those rules apply. It says &quot;When
checking<br>
&gt; for glare conditions&quot;. That is too vague for me. Under what<br>
&gt; circumstances does the UA check for glare conditions?</font></tt>
<br>
<br><tt><font size=2>Yes. I also think the words is vague. </font></tt>
<br><tt><font size=2>So, I think we may need *words enhancement*, if my
understanding of it's technical semantics is correct.</font></tt>
<br><tt><font size=2>If it's technical semantics is not correct or not
comprehensive enough, we might need *words and technical enhancement* at
the same time here.</font></tt>
<br>
<br><tt><font size=2>My point here is that as ther's arguement about it's
technical semantics's correctness(IMO, section 3.5 and 4.5 is correct,
technically), it would be more acceptable for me to describe it as &quot;its
an *potential* undesirable change too&quot;.</font></tt>
<br><tt><font size=2><br>
&gt; <br>
&gt; My longer message on glare, etc. is probably a better context within<br>
&gt; which to have this discussion.</font></tt>
<br>
<br><tt><font size=2>I am reading it, and will go on soon.</font></tt>
<br>
<br><tt><font size=2>Thanks,</font></tt>
<br>
<br><tt><font size=2>Gao<br>
</font></tt><br><pre>
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From shinji.okumura@softfront.jp  Sun May 30 21:22:02 2010
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From: OKUMURA Shinji <shinji.okumura@softfront.jp>
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Gao,

>> I understand.
>> 
>> The text says that the UAC can not expect to receive a 487
>> response to the INVITE. Right?
>
>As UAC's old contact has been lost, UAC can not expect to receive any 
>final response for the old Re-INVITE here.
>So, the specific for final response is not important here. And the UAC's 
>next Re-INVITE as synchronization can refresh the state here. 

Even if the UAS would send a new UPDATE to undo the changes
and return a 2xx to the reINVITE, the UAC never receive the 2xx.
The UAC will assume that the reINVITE is rejected with 408
(Request Timeout) and send a new UPDATE or reINVITE to
synchronize a common view.
And if the UAS responds 2xx to the reINVITE, it waits for an
ACK from the UAC, but the UAC never send an ACK.
If the UAS responds a 487, it can expect to receive an ACK
from an adjacent proxy.

After all, if a UAS receives a target-refresh UPDATE within
a reINVITE transaction, the UAS should send a 4xx(487?) to
the reINVITE immediately without sending a new UPDATE to
undo the changes.

>> And now, assuming that some of changes have already been
>> executed, the UAS would send a new UPDATE to undo the changes.
>> But the local target of the UAC has been changed, so I think UAC
>> should an UPDATE to announce own new target before sending the
>> CANCEL.
>> 
>> Do you think so?
>
>I think the UAC should send a target refresh UPDATE before CANCEL. But the 
>reason is not about UAS's rollback usage UPDATE.(the state would be 
>synchronized by a next Re-INVITE).
>
>To send a new request(the CANCEL) at new address, sending a target refresh 
>UPDATE before the new request (the CANCEL) might be better.

IMO a CANCEL transaction doesn't depend on a local target.
It has only one Via header that is identical to the one of
the reINVITE.

The UAC may not be able to receive a response to the CANCEL
and an adjacent proxy may retransmit the response.

Should the UAC send the CANCEL?

Regards,
Shinji

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> >As UAC's old contact has been lost, UAC can not expect to receive any 
> >final response for the old Re-INVITE here.
> >So, the specific for final response is not important here. And the 
UAC's 
> >next Re-INVITE as synchronization can refresh the state here. 
> 
> Even if the UAS would send a new UPDATE to undo the changes
> and return a 2xx to the reINVITE, the UAC never receive the 2xx.
> The UAC will assume that the reINVITE is rejected with 408
> (Request Timeout) and send a new UPDATE or reINVITE to
> synchronize a common view.

I think, a new UPDATE or ReINVITE to synchronize state is mandatory here. 
I guess you have the same point here.

4xx to the old ReINVITE is not proper, if there's some change/modification 
in the ReINVITE has be enforced. This is the basic concept in the draft.

If there's no change/modification in the ReINVITE has be enforced, 4xx 
would be OK.

> And if the UAS responds 2xx to the reINVITE, it waits for an
> ACK from the UAC, but the UAC never send an ACK.
> If the UAS responds a 487, it can expect to receive an ACK
> from an adjacent proxy.
> 
> After all, if a UAS receives a target-refresh UPDATE within
> a reINVITE transaction, the UAS should send a 4xx(487?) to
> the reINVITE immediately without sending a new UPDATE to
> undo the changes.
> 
> >> And now, assuming that some of changes have already been
> >> executed, the UAS would send a new UPDATE to undo the changes.
> >> But the local target of the UAC has been changed, so I think UAC
> >> should an UPDATE to announce own new target before sending the
> >> CANCEL.
> >> 
> >> Do you think so?
> >
> >I think the UAC should send a target refresh UPDATE before CANCEL. But 
the 
> >reason is not about UAS's rollback usage UPDATE.(the state would be 
> >synchronized by a next Re-INVITE).
> >
> >To send a new request(the CANCEL) at new address, sending a target 
refresh 
> >UPDATE before the new request (the CANCEL) might be better.
> 
> IMO a CANCEL transaction doesn't depend on a local target.

Yes. But if the CANCEL send before the target refresh, UAC would not get 
the response.
That's why I think a target refresh request would be better. But this is 
not the critical point here.
So, if there's vendors want to send CANCEL before target refresh UPDATE, I 
am OK with it.

> It has only one Via header that is identical to the one of
> the reINVITE.
> 
> The UAC may not be able to receive a response to the CANCEL
> and an adjacent proxy may retransmit the response.
> 
> Should the UAC send the CANCEL?

You seems arguing using target refresh UPDATE to trigger UAS's final 
response. IMO, CANCEL might be better here.

And it is more natural for UAC to send a CANCEL, if it want to terminate 
current pending ReINVITE.


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<br><tt><font size=2><br>
&gt; &gt;As UAC's old contact has been lost, UAC can not expect to receive
any <br>
&gt; &gt;final response for the old Re-INVITE here.<br>
&gt; &gt;So, the specific for final response is not important here. And
the UAC's <br>
&gt; &gt;next Re-INVITE as synchronization can refresh the state here.
<br>
&gt; <br>
&gt; Even if the UAS would send a new UPDATE to undo the changes<br>
&gt; and return a 2xx to the reINVITE, the UAC never receive the 2xx.<br>
&gt; The UAC will assume that the reINVITE is rejected with 408<br>
&gt; (Request Timeout) and send a new UPDATE or reINVITE to<br>
&gt; synchronize a common view.</font></tt>
<br>
<br><tt><font size=2>I think, a new UPDATE or ReINVITE to synchronize state
is mandatory here. I guess you have the same point here.</font></tt>
<br>
<br><tt><font size=2>4xx to the old ReINVITE is not proper, if there's
some change/modification in the ReINVITE has be enforced. This is the basic
concept in the draft.</font></tt>
<br>
<br><tt><font size=2>If there's no change/modification in the ReINVITE
has be enforced, 4xx would be OK.</font></tt>
<br><tt><font size=2><br>
&gt; And if the UAS responds 2xx to the reINVITE, it waits for an<br>
&gt; ACK from the UAC, but the UAC never send an ACK.<br>
&gt; If the UAS responds a 487, it can expect to receive an ACK<br>
&gt; from an adjacent proxy.<br>
&gt; <br>
&gt; After all, if a UAS receives a target-refresh UPDATE within<br>
&gt; a reINVITE transaction, the UAS should send a 4xx(487?) to<br>
&gt; the reINVITE immediately without sending a new UPDATE to<br>
&gt; undo the changes.<br>
&gt; <br>
&gt; &gt;&gt; And now, assuming that some of changes have already been<br>
&gt; &gt;&gt; executed, the UAS would send a new UPDATE to undo the changes.<br>
&gt; &gt;&gt; But the local target of the UAC has been changed, so I think
UAC<br>
&gt; &gt;&gt; should an UPDATE to announce own new target before sending
the<br>
&gt; &gt;&gt; CANCEL.<br>
&gt; &gt;&gt; <br>
&gt; &gt;&gt; Do you think so?<br>
&gt; &gt;<br>
&gt; &gt;I think the UAC should send a target refresh UPDATE before CANCEL.
But the <br>
&gt; &gt;reason is not about UAS's rollback usage UPDATE.(the state would
be <br>
&gt; &gt;synchronized by a next Re-INVITE).<br>
&gt; &gt;<br>
&gt; &gt;To send a new request(the CANCEL) at new address, sending a target
refresh <br>
&gt; &gt;UPDATE before the new request (the CANCEL) might be better.<br>
&gt; <br>
&gt; IMO a CANCEL transaction doesn't depend on a local target.</font></tt>
<br>
<br><tt><font size=2>Yes. But if the CANCEL send before the target refresh,
UAC would not get the response.</font></tt>
<br><tt><font size=2>That's why I think a target refresh request would
be better. But this is not the critical point here.</font></tt>
<br><tt><font size=2>So, if there's vendors want to send CANCEL before
target refresh UPDATE, I am OK with it.</font></tt>
<br><tt><font size=2><br>
&gt; It has only one Via header that is identical to the one of<br>
&gt; the reINVITE.<br>
&gt; <br>
&gt; The UAC may not be able to receive a response to the CANCEL<br>
&gt; and an adjacent proxy may retransmit the response.<br>
&gt; <br>
&gt; Should the UAC send the CANCEL?</font></tt>
<br>
<br><tt><font size=2>You seems arguing using target refresh UPDATE to trigger
UAS's final response. IMO, CANCEL might be better here.</font></tt>
<br>
<br><tt><font size=2>And it is more natural for UAC to send a CANCEL, if
it want to terminate current pending ReINVITE.</font></tt>
<br><br><pre>
--------------------------------------------------------
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From shinji.okumura@softfront.jp  Sun May 30 23:10:11 2010
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From: OKUMURA Shinji <shinji.okumura@softfront.jp>
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Date: Mon, 31 May 2010 15:09:56 +0900
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare andUPDATE	withoutSDP
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Paul,

Paul Kyzivat <pkyzivat@cisco.com>
Sun, 30 May 2010 18:28:29 -0400
>Shinji,
>
>I agree with what you say, but perhaps for different reasons, for based 
>on different view of what "glare" means.
>
>IMO the 491 response should be used to indicate glare has been detected, 
>specifically that a proposal to negotiate a state change has been 
>*received* while a previously *sent* proposal to negotiate a state 
>change is still unresolved. That can apply to o/a regardless of what 
>messages cause the situation to be detected. And I think this can apply 
>even when receiving an offerless reINVITE, which is a demand for an offer.
>
>OTOH the 500 response applies when a proposal to negotiate a state 
>change has been *received* while a previously *received* proposal to 
>negotiate state is still unresolved. This is not glare, because the 
>problem is not the result of coincidence in valid behavior. The 500 is a 
>suitable response because it is bad behavior by the sender of those 
>proposals that results in the problem.

IMO whether your view and mine are different depend on one
disputed point.

Which is a UAC that sends an offerless reINVITE, a "sender" of
the negotiation or a "receiver"?

If it is a "sender", then we say the same thing by a different
wording. I call it "client transaction", you call it "sender".

You say "an offerless reINVITE is a demand for an offer".
I think we are agreeing.

Regards,
Shinji

>	Thanks,
>	Paul
>
>OKUMURA Shinji wrote:
>> Hi Brett,
>> 
>> I have a similar question too.
>> 
>> RFC 3261 and 3311 describe the definition regarding 500 and
>> 491 response.
>> When UA rejects the INVITE during an INVITE transaction, or
>> rejects the UPDATE during an UPDATE transaction, it has nothing
>> to do with offer-answer which error code to be used.
>> 
>> If both transaction have the same direction, the error code is 500.
>> If not, the error code is 491.
>> 
>> But an interworking of INVITE and UPDATE is unclear.
>> The draft-offeranswer makes an efforts to clarify it in the
>> section 4.3.
>> 
>>> Howdy,
>>>
>>> Draft-ietf-sipcore-reinvite-04 sections 3.5 and 4.5 indicate
>>> that an UPDATE without SDP should be treated (from a glare
>>> perspective) as though it contained an offer SDP.
>>> However I currently find the definition of "glare" and the
>>> expected behavior to be ambiguous.
>>>
>>> What is the precise meaning of glare?  I ask mainly because
>>> some vendors send a 491 when an RFC indicates to send a 500.
>>>
>>> If INVITE contains offer SDP and 18x was not reliable, how
>>> SHOULD the called party respond if UPDATE without SDP is
>>> received from caller: 491, 500, or process as though no glare?
>> 
>> If UA must refuse it , parhaps it is 500.
>> IMO it is another question whether UA must reject the UPDATE
>> without an offer.
>> 
>>> If INVITE contains offer SDP and 18x was not reliable, how
>>> SHOULD the caller respond if UPDATE without SDP is received
>>>from called party: 491, 500, or process as though no glare?
>> 
>> Parhaps it is 491.
>> 
>>> If caller receives INVITE 2xx without offer/answer, how SHOULD
>>> or MUST the caller respond if UPDATE without SDP is received
>>>from called party prior to sending ACK (if not willing to
>>> queue or drop UPDATE): 491 or 500?  Is it a SHOULD or MUST?
>> 
>> UA may send 2xx to UPDATE. Because the INVITE transaction has
>> been completed.
>> 
>> For reference, the following figures are now under consideration.
>> 
>>   A                      B
>>   |                      |
>>   |INVITE                |
>>   |--------------------->|<-+             <-----+
>>   |  1xx(offer or answer)|  |                   |
>>   |<---------------------|  | reject INVITE     | reject UPDATE
>>   |PRACK                 |  | Don't send INVITE | Don't send UPDATE
>>   |--------------------->|  |                   |
>>   |               2xx-PRA|  |                   |
>>   |<---------------------|  |             <-----+
>>   |                      |  |                   | may send UPDATE
>>   |                   2xx|  |                   | may accept UPDATE
>>   |<---------------------|<-+                   |
>>   |                      |  | may send INVITE   |
>>   |ACK                   |  | may accept INVITE |
>>   |--------------------->|  |                   |
>>   |                      |  v                   v
>>   |                      |
>>           Figure 16
>> 
>>   A                      B
>>   |                      |
>>   |INVITE                |
>>   |--------------------->|<-+             <-----+
>>   |  1xx(offer or answer)|  |                   |
>>   |<---------------------|  | reject INVITE     | reject UPDATE
>>   |PRACK                 |  | Don't send INVITE | Don't send UPDATE
>>   |--------------------->|  |                   |
>>   |               2xx-PRA|  |                   |
>>   |<---------------------|  |             <-----+
>>   |                      |  |                   | may send UPDATE
>>   |                   4xx|  |                   | may acceptUPDATE
>>   |<---------------------|<-+                   |
>>   |                      |  | Don't send INVITE |
>>   |ACK          ACK      |  | may accept INVITE | <-- Should B reject UPDATE or not?
>>   |--------->   -------->|<-+             <-----+     That is another question.
>>   |                      |  |                   |
>>   |                      |  v                   v
>>           Figure 17
>> 
>> In both figures, 
>> UA A reject all INVITE/UPDATE with 491.
>> UA B reject all INVITE/UPDATE with 500.
>> 
>> Regards,
>> Shinji
>
>_______________________________________________
>sipcore mailing list
>sipcore@ietf.org
>https://www.ietf.org/mailman/listinfo/sipcore

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From: OKUMURA Shinji <shinji.okumura@softfront.jp>
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE	without SDP
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Paul,

In my recognition, Gao and I are not agreeing.

I have two comments regarding Figure 2.

1. Between a non-2xx to a reINVITE and an ACK, the UAS may
   accept a receiving UPDATE.

   This draft seems to forbid the UPDATE.

2. In this case, the common view of both UAs has already
   been synchronized. IMO if an UPDATE has been not used in
   offer-answer exchange before 4xx, then the original
   rollback process may work well.
   That is, a following UPDATE is unnecessary in this case.

Does Gao agree with my comments?

Regards,
Shinji

Paul Kyzivat <pkyzivat@cisco.com>
Sun, 30 May 2010 18:16:31 -0400
>I read through this discussion, and I cannot figure out what the 
>difference is between the two of you. As of this reply it seems to me 
>that the two of you are agreeing.
>
>Can one of you restate the issue and the contrasting solutions?
>
>	Thanks,
>	Paul
>
>OKUMURA Shinji wrote:
>> Gao,
>> 
>> This is a figure I've assumed.
>> 
>>     A                               B
>>     |                               |
>>   s0|re-INV (offer1)                |s0
>>     |------------------------------>|
>>     |               18x-rel(answer1)|
>>     |<------------------------------|
>>   s1|                        non-2xx|s1
>>     |<------------------------------|<-+
>>   s0|ACK                            |s0|
>>     |----------->                   |  |
>>     |UPD(offer0)                    |  |
>>     |==============================>|* | reject UPDATE
>>     |                       491(UPD)|  |
>>     |<==============================|  |
>>     |                    ACK        |  |
>>     |                    ---------->|<-+
>>     |                               |
>>                 Figure 2
>> 
>>> Hi,
>>>
>>> OKUMURA Shinji <shinji.okumura@softfront.jp>
>>>> Gao,
>>>>
>>>> gao.yang2@zte.com.cn
>>>> Tue, 25 May 2010 14:15:10 +0800
>>>>> Hi,
>>>>>
>>>>>>>> I agree that it doesn't forbid sending it.  It allows it and 
>>>>>>>> mandates the UAS return 491 or 500 (depending upon direction).
>>>>>>> It mandates the UAS send 4xx/5xx to UPDATE only when the UAS
>>>>>>> send 4xx/5xx to the Re-INVITE and the UAS get the UPDATE before
>>>>>>> ACK of 4xx/5xx(Re-INVITE).
>>>>>> As you know, ACK to non-2xx is hop-by-hop.
>>>>>> There is little benefit to reject such an UPDATE.
>>>>> UAS send 491 for the UPDATE(in the flow) to *tell* UAC that in it's
>>>>> (UAS) 
>>>>> view, there's a glare case.
>>>> But this UPDATE is sent in order to synchronize the common view.
>>>> If UAS return 2xx, then the situation goes into nonglare.
>>> I guess I have a inverse point here. By this draft's clarification, 
>>> there's no inconsistent session state for the non-successful Re-INVITE. 
>>> So, why UAC send a UPDATE to synchronize the common view?
>> 
>> In 3.4.  UAC Behavior,
>>   "In order to cope with these race condition
>>    situations, a UAC that receives an error response to a re-INVITE for
>>    which changes have been already executed SHOULD generate a new re-
>>    INVITE or UPDATE request in order to make sure that both UAs have a
>>    common view of the state of the dialog and the session."
>> 
>> Under the above descriptions, this UAC generates a new UPDATE.
>> 
>>> While if UAS return 2xx(for the UPDATE), the situation would get into 
>>> glare(UAS has no way to figure out this UPDATE is in-scope of Re-INVITE or 
>>> not). This is relating to atomic of session state. I'd like to emphasize 
>>> that keeping Re-INVITE atomic, except for target refresh, is the main idea 
>>> in this draft.
>> 
>> This issue has nothing to do with the "atomic of session state".
>> 
>>>> Probably another UPDATE will be sent, but this UPDATE
>>>> transaction is supposed to be avoided.
>>>>
>>>> At the very start, the UAS rejects a reINVITE does not care
>>>> about this text. Is it necessary to let such a UAS follow a
>>>> new rule?
>>> These days, you guys showed a lot of use cases. I don't remember the 
>>> details. Could you say more about this use case.
>> 
>> Please look at the original mail of mine, and re-comment it.
>> You seem to be confused a little.
>> 
>>>>> When UAS get the ACK then UPDATE, it still has no precise criterion for 
>>>>> whether the UAC gets non-2xx(Re-INVITE) first then send the UPDATE or not. 
>>>>> Keeping rejecting the UPDATE is not a reasonable way here. So, the draft 
>>>>> suggest the way, as UAS can send 2xx to the UPDATE.
>>>> It is a conventional handling and enough for UAS.
>>>>
>>>>> When UAC get UAS's 4xx of UPDATE, the session state is clear for it. And 
>>>>> UAC would only send a synchronizing Re-INVITE/UPDATE when it is a glare 
>>>>> case in the UAC's view.
>>>> UAS' behavior does not affect UAC' behavior directly.
>>>> UAC will do the needful regardless of whether UAS rejects
>>>> a reINVITE.
>>>>
>>>>> IMO, this way is better than the way we might once use: allowing B send
>>>>> 2xx to UPDATE, and both A and B *see* glare, and all send Re-INVITE/UPDATE 
>>>>> to refresh session state by each's view.
>>>> The new rule for UAS does not seem to improve the situation.
>>>> The new rule for UAC seems to be just enough to synchronize
>>>> the view.
>>> The key merit for the new rule is make both UAC and UAS clear in which 
>>> situation is it's responsibility to handle glare case, and there's only 
>>> one UA need to do something. Both sides want to refresh session state by 
>>> each's view is not a elegant way.
>> 
>> The crux of the matter is that a UAS can not recognize own glare
>> situation anytime.
>> It is UAC's duty to synchronize a common view.
>> 
>> Regards,
>> Shinji
>> 
>>> Thanks,
>>>
>>> Gao

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From: OKUMURA Shinji <shinji.okumura@softfront.jp>
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare andUPDATE	withoutSDP
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Paul,

In RFC3311 5.2 Receiving an UPDATE
   A UAS that receives an UPDATE before it has generated a final
   response to a previous UPDATE on the same dialog MUST return a 500
   response to the new UPDATE, and MUST include a Retry-After header
   field with a randomly chosen value between 0 and 10 seconds.

The above descriptions have nothing to do with an offer-answer.

Should this rule be applied only to an offer-answer situation?

Regards,
Shinji

Paul Kyzivat <pkyzivat@cisco.com>
Sun, 30 May 2010 18:28:29 -0400
>Shinji,
>
>I agree with what you say, but perhaps for different reasons, for based 
>on different view of what "glare" means.
>
>IMO the 491 response should be used to indicate glare has been detected, 
>specifically that a proposal to negotiate a state change has been 
>*received* while a previously *sent* proposal to negotiate a state 
>change is still unresolved. That can apply to o/a regardless of what 
>messages cause the situation to be detected. And I think this can apply 
>even when receiving an offerless reINVITE, which is a demand for an offer.
>
>OTOH the 500 response applies when a proposal to negotiate a state 
>change has been *received* while a previously *received* proposal to 
>negotiate state is still unresolved. This is not glare, because the 
>problem is not the result of coincidence in valid behavior. The 500 is a 
>suitable response because it is bad behavior by the sender of those 
>proposals that results in the problem.
>
>	Thanks,
>	Paul
>
>OKUMURA Shinji wrote:
>> Hi Brett,
>> 
>> I have a similar question too.
>> 
>> RFC 3261 and 3311 describe the definition regarding 500 and
>> 491 response.
>> When UA rejects the INVITE during an INVITE transaction, or
>> rejects the UPDATE during an UPDATE transaction, it has nothing
>> to do with offer-answer which error code to be used.
>> 
>> If both transaction have the same direction, the error code is 500.
>> If not, the error code is 491.
>> 
>> But an interworking of INVITE and UPDATE is unclear.
>> The draft-offeranswer makes an efforts to clarify it in the
>> section 4.3.
>> 
>>> Howdy,
>>>
>>> Draft-ietf-sipcore-reinvite-04 sections 3.5 and 4.5 indicate
>>> that an UPDATE without SDP should be treated (from a glare
>>> perspective) as though it contained an offer SDP.
>>> However I currently find the definition of "glare" and the
>>> expected behavior to be ambiguous.
>>>
>>> What is the precise meaning of glare?  I ask mainly because
>>> some vendors send a 491 when an RFC indicates to send a 500.
>>>
>>> If INVITE contains offer SDP and 18x was not reliable, how
>>> SHOULD the called party respond if UPDATE without SDP is
>>> received from caller: 491, 500, or process as though no glare?
>> 
>> If UA must refuse it , parhaps it is 500.
>> IMO it is another question whether UA must reject the UPDATE
>> without an offer.
>> 
>>> If INVITE contains offer SDP and 18x was not reliable, how
>>> SHOULD the caller respond if UPDATE without SDP is received
>>>from called party: 491, 500, or process as though no glare?
>> 
>> Parhaps it is 491.
>> 
>>> If caller receives INVITE 2xx without offer/answer, how SHOULD
>>> or MUST the caller respond if UPDATE without SDP is received
>>>from called party prior to sending ACK (if not willing to
>>> queue or drop UPDATE): 491 or 500?  Is it a SHOULD or MUST?
>> 
>> UA may send 2xx to UPDATE. Because the INVITE transaction has
>> been completed.
>> 
>> For reference, the following figures are now under consideration.
>> 
>>   A                      B
>>   |                      |
>>   |INVITE                |
>>   |--------------------->|<-+             <-----+
>>   |  1xx(offer or answer)|  |                   |
>>   |<---------------------|  | reject INVITE     | reject UPDATE
>>   |PRACK                 |  | Don't send INVITE | Don't send UPDATE
>>   |--------------------->|  |                   |
>>   |               2xx-PRA|  |                   |
>>   |<---------------------|  |             <-----+
>>   |                      |  |                   | may send UPDATE
>>   |                   2xx|  |                   | may accept UPDATE
>>   |<---------------------|<-+                   |
>>   |                      |  | may send INVITE   |
>>   |ACK                   |  | may accept INVITE |
>>   |--------------------->|  |                   |
>>   |                      |  v                   v
>>   |                      |
>>           Figure 16
>> 
>>   A                      B
>>   |                      |
>>   |INVITE                |
>>   |--------------------->|<-+             <-----+
>>   |  1xx(offer or answer)|  |                   |
>>   |<---------------------|  | reject INVITE     | reject UPDATE
>>   |PRACK                 |  | Don't send INVITE | Don't send UPDATE
>>   |--------------------->|  |                   |
>>   |               2xx-PRA|  |                   |
>>   |<---------------------|  |             <-----+
>>   |                      |  |                   | may send UPDATE
>>   |                   4xx|  |                   | may acceptUPDATE
>>   |<---------------------|<-+                   |
>>   |                      |  | Don't send INVITE |
>>   |ACK          ACK      |  | may accept INVITE | <-- Should B reject UPDATE or not?
>>   |--------->   -------->|<-+             <-----+     That is another question.
>>   |                      |  |                   |
>>   |                      |  v                   v
>>           Figure 17
>> 
>> In both figures, 
>> UA A reject all INVITE/UPDATE with 491.
>> UA B reject all INVITE/UPDATE with 500.
>> 
>> Regards,
>> Shinji
>
>_______________________________________________
>sipcore mailing list
>sipcore@ietf.org
>https://www.ietf.org/mailman/listinfo/sipcore

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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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> I *think* that considerations of *glare* occur when there is something 
> to be *negotiated* and either side may initiate the negotiation. The 
> negotiated piece of state that has been considered to date for glare is 
> the pair of SDP bodies that describe the media sessions.

Your classification: 

"O/A glare case" and "Session Timer glare case" are subset of "negotiated 
glare case";

"Target-refresh ambiguous misorder" and "Updating PAI" are subset of 
"Declaring ambiguous misorder";

"negotiated glare case" and "Declaring ambiguous misorder" are subset of 
"ambiguous case".

I once had a classification as "glare case" = "ambiguous case". But if we 
choose the "Patching way"(see below), I guess my classification is enough. 
If we want the "Refashion way", your more refined classification might be 
needed.

> 
> As noted elsewhere in this thread, another piece state that is 
> *negotiated* is session-timer. As a result, there could be a glare 
> situation around that. The determination of whether glare has occurred 
> ought to be independent for each kind of negotiated state. So one 
> determination for SDP, another for session-timer. But there is only one 
> mechanism for signaling the glare in either case, the 491 response code. 

> So it would be used if *either* sdp-glare or session-timer-glare 
occurred.
> 
> Target-refresh is somewhat different, because it isn't *negotiated* - it 

> is simply *declared* independently in each direction. So I don't think 
> it really ought to be considered a glare situation.
> 
> Another in this category is sending an updated PAI, or changing From. 
> This to is a declaration rather than a negotiation, so that glare 
> shouldn't apply.
> 
> In both of those cases, there remains an ordering issue, and perhaps the 

> answer is to forbid behavior that can result in ambiguity. But I fear 
>that cure is worse than the disease, so I'm inclined to seek a better 
cure.

I think there are two different methodology for this issue:
1. Patching way: Saving current right case as it is, and cure ambiguous 
case without any breach for current right case(or making the breach as 
little as possible). I think the way defined in the draft belongs to 
"Patching way". 

2. Refashion way: Based on the refined classification of "ambiguous case", 
seek solution and redefine rules on each classification's semantics. This 
way might forbid some *current right case*.

Technically, I think "Refashion way" is better. But considering there's 
would be a serious interworking issue behind the "Refashion way"(there 
would be a quit long period for most vendors adopting this draft), I 
finally prefer the "Patching way" for nowadays.

Personally, I prefer this draft has relative narrow aspect, and with tiny 
impact on current right working things. If someday, we have a 3261bis 
plan, I would prefer the "Refashion way" more.

> 
> I'm especially concerned about that in the case of target refresh. If 
> the sender has lost its old address, then we are on the brink of losing 
> the call entirely. Saving the call is more important than whether we 
> have the proper o/a state - that can be fixed after the new target 
> address is established. 

I admire this point.

> Also, target refresh cases are typically rare. 
> Two successive ones in close succession in the same direction should be 
> rare^2. (A target refresh of both ends in quick succession could be more 

> likely - for instance if the same event causes both to lose their 
> address. It seems unlikely that this will be a recoverable event.)
> 
> So, I think we ought not be forbidding target refresh just because of 
> other state. Ambiguity about which is the old target and which is the 
> new one can be measured by which is a change - assuming there aren't two 

> changes in quick succession. I think Gao has also recommended this.

Yes. And I think Gonzalo's current text in the draft covered this, 
technically. 

While some words level enhancement might needed for enhancing it's 
understandability. 

Thanks,

Gao


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<br><tt><font size=2><br>
<br>
<br>
&gt; I *think* that considerations of *glare* occur when there is something
<br>
&gt; to be *negotiated* and either side may initiate the negotiation. The
<br>
&gt; negotiated piece of state that has been considered to date for glare
is <br>
&gt; the pair of SDP bodies that describe the media sessions.</font></tt>
<br>
<br><tt><font size=2>Your classification: </font></tt>
<br>
<br><tt><font size=2>&quot;O/A glare case&quot; and &quot;Session Timer
glare case&quot; are subset of &quot;negotiated glare case&quot;;</font></tt>
<br>
<br><tt><font size=2>&quot;Target-refresh ambiguous misorder&quot; and
&quot;Updating PAI&quot; are subset of &quot;Declaring ambiguous misorder&quot;;</font></tt>
<br>
<br><tt><font size=2>&quot;negotiated glare case&quot; and &quot;Declaring
ambiguous misorder&quot; are subset of &quot;ambiguous case&quot;.</font></tt>
<br>
<br><tt><font size=2>I once had a classification as &quot;glare case&quot;
= &quot;ambiguous case&quot;. But if we choose the &quot;Patching way&quot;(see
below), I guess my classification is enough. If we want the &quot;Refashion
way&quot;, your more refined classification might be needed.</font></tt>
<br><tt><font size=2><br>
&gt; <br>
&gt; As noted elsewhere in this thread, another piece state that is <br>
&gt; *negotiated* is session-timer. As a result, there could be a glare
<br>
&gt; situation around that. The determination of whether glare has occurred
<br>
&gt; ought to be independent for each kind of negotiated state. So one
<br>
&gt; determination for SDP, another for session-timer. But there is only
one <br>
&gt; mechanism for signaling the glare in either case, the 491 response
code. <br>
&gt; So it would be used if *either* sdp-glare or session-timer-glare occurred.<br>
&gt; <br>
&gt; Target-refresh is somewhat different, because it isn't *negotiated*
- it <br>
&gt; is simply *declared* independently in each direction. So I don't think
<br>
&gt; it really ought to be considered a glare situation.<br>
&gt; <br>
&gt; Another in this category is sending an updated PAI, or changing From.
<br>
&gt; This to is a declaration rather than a negotiation, so that glare
<br>
&gt; shouldn't apply.<br>
&gt; <br>
&gt; In both of those cases, there remains an ordering issue, and perhaps
the <br>
&gt; answer is to forbid behavior that can result in ambiguity. But I fear
</font></tt>
<br><tt><font size=2>&gt;that cure is worse than the disease, so I'm inclined
to seek a better cure.</font></tt>
<br>
<br><tt><font size=2>I think there are two different methodology for this
issue:</font></tt>
<br><tt><font size=2>1. Patching way: Saving current right case as it is,
and cure ambiguous case without any breach for current right case(or making
the breach as little as possible). I think the way defined in the draft
belongs to &quot;Patching way&quot;. </font></tt>
<br>
<br><tt><font size=2>2. Refashion way: Based on the refined classification
of &quot;ambiguous case&quot;, seek solution and redefine rules on each
classification's semantics. This way might forbid some *current right case*.</font></tt>
<br>
<br><tt><font size=2>Technically, I think &quot;Refashion way&quot; is
better. But considering there's would be a serious interworking issue behind
the &quot;Refashion way&quot;(there would be a quit long period for most
vendors adopting this draft), I finally prefer the &quot;Patching way&quot;
for nowadays.</font></tt>
<br>
<br><tt><font size=2>Personally, I prefer this draft has relative narrow
aspect, and with tiny impact on current right working things. If someday,
we have a 3261bis plan, I would prefer the &quot;Refashion way&quot; more.</font></tt>
<br><tt><font size=2><br>
&gt; <br>
&gt; I'm especially concerned about that in the case of target refresh.
If <br>
&gt; the sender has lost its old address, then we are on the brink of losing
<br>
&gt; the call entirely. Saving the call is more important than whether
we <br>
&gt; have the proper o/a state - that can be fixed after the new target
<br>
&gt; address is established. </font></tt>
<br>
<br><tt><font size=2>I admire this point.</font></tt>
<br>
<br><tt><font size=2>&gt; Also, target refresh cases are typically rare.
<br>
&gt; Two successive ones in close succession in the same direction should
be <br>
&gt; rare^2. (A target refresh of both ends in quick succession could be
more <br>
&gt; likely - for instance if the same event causes both to lose their
<br>
&gt; address. It seems unlikely that this will be a recoverable event.)<br>
&gt; <br>
&gt; So, I think we ought not be forbidding target refresh just because
of <br>
&gt; other state. Ambiguity about which is the old target and which is
the <br>
&gt; new one can be measured by which is a change - assuming there aren't
two <br>
&gt; changes in quick succession. I think Gao has also recommended this.</font></tt>
<br>
<br><tt><font size=2>Yes. And I think Gonzalo's current text in the draft
covered this, technically. </font></tt>
<br>
<br><tt><font size=2>While some words level enhancement might needed for
enhancing it's understandability. </font></tt>
<br>
<br><tt><font size=2>Thanks,</font></tt>
<br>
<br><tt><font size=2>Gao</font></tt>
<br><br><pre>
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To: "sipcore@ietf.org" <sipcore@ietf.org>
Date: Mon, 31 May 2010 12:06:11 +0200
Thread-Topic: Draft new version: draft-ietf-sipcore-keep-04
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Subject: [sipcore] Draft new version: draft-ietf-sipcore-keep-04
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Hi,

Based on the discussions and comments, I have submitted a new version of th=
e keep draft now.

There are currently no open issues, and in my opinion the draft is ready fo=
r WGLC.

Regards,

Christer


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<div>&nbsp;</div>
<div><font size=3D"2">Hi,</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Based on the discussions and comments, I have submitt=
ed a new version of the keep draft now.</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">There are currently no open issues, and in my opinion=
 the draft is ready for WGLC.</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Regards,</font></div>
<div><font size=3D"2">&nbsp;</font></div>
<div><font size=3D"2">Christer</font></div>
<div><font size=3D"2">&nbsp;</font></div>
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From christer.holmberg@ericsson.com  Mon May 31 03:11:53 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@cisco.com>, "gao.yang2@zte.com.cn" <gao.yang2@zte.com.cn>
Date: Mon, 31 May 2010 12:11:38 +0200
Thread-Topic: [sipcore] REg. draft-ietf-sipcore-reinvite-04  //Session Timer
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References: <OFAD8B7FED.AAEB71E6-ON4825772D.0023D33D-4825772D.0025D4A9@zte.com.cn> <4C02D9E0.5060306@cisco.com>
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Cc: "sipcore@ietf.org" <sipcore@ietf.org>, Harbhanu <harbhanu@huawei.com>
Subject: Re: [sipcore] REg. draft-ietf-sipcore-reinvite-04  //Session Timer
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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-04.txt
	Pages           : 15
	Date            : 2010-05-31

This specification defines a new Session Initiation Protocol (SIP)
Via header field parameter, "keep", which allows adjacent SIP
entities to explicitly negotiate usage of the Network Address
Translation (NAT) keep-alive mechanisms defined in SIP Outbound, in
cases where SIP Outbound is not supported, cannot be applied, or
where usage of keep-alives is not implicitly negotiated as part of
the SIP Outbound negotiation.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sipcore-keep-04.txt

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From shinji.okumura@softfront.jp  Mon May 31 03:17:09 2010
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From: OKUMURA Shinji <shinji.okumura@softfront.jp>
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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Paul,

Paul Kyzivat <pkyzivat@cisco.com>
Sun, 30 May 2010 17:25:37 -0400
>Sorry to be so slow in responding. I got behind, and had to wait till I 
>had time to read through all that has been said before I could make any 
>sort of meaningful response.
>
>As it is, there is so much that I am responding after reading most of 
>the -reinvite- traffic, without first checking if some of the mail with 
>different subject is also relevant.
>
>gao.yang2@zte.com.cn wrote:
>> 
>> Rules for "When checking for glare situations" in the draft text means 
>> the UAs's reasoning and handling while the UAs are in a glare 
>> situations, which is ambiguous.
>> 
>> It does not means UAs reasoning and handling for every use case to 
>> figure out whether this case is a glare case.
>> 
>> This is my understanding. And I'd like other co-authors and important 
>> reviewer to comment on this point.
>
>In light of all the discussion this has generated, I think -reinvite- is 
>not sufficiently clear on what it means. I think we need to be looking 
>for a way to revise it to be more clear, regardless of its current 
>post-WGLC status. (Note that -offeranswer- is also in post-WGLC status, 
>and we are debating changes to it.)
>
>We *REALLY* need to hear from Gonzalo for his opinion on this!
>
>My opinion at this point is that stating that "UPDATEs without SDP 
>should be treated as if they contained an offer" is an insufficient 
>technique for stating the required behavior - that it needs to be stated 
>more explicitly, even though that will take more words.
>
>I *think* that considerations of *glare* occur when there is something 
>to be *negotiated* and either side may initiate the negotiation. The 
>negotiated piece of state that has been considered to date for glare is 
>the pair of SDP bodies that describe the media sessions.
>
>As noted elsewhere in this thread, another piece state that is 
>*negotiated* is session-timer. As a result, there could be a glare 
>situation around that. The determination of whether glare has occurred 
>ought to be independent for each kind of negotiated state. So one 
>determination for SDP, another for session-timer. But there is only one 
>mechanism for signaling the glare in either case, the 491 response code. 
>So it would be used if *either* sdp-glare or session-timer-glare occurred.
>
>Target-refresh is somewhat different, because it isn't *negotiated* - it 
>is simply *declared* independently in each direction. So I don't think 
>it really ought to be considered a glare situation.

Exactly!
I've thought the differences between offer-answer and target-
refresh as below,

1. When a UAS sends 4xx to reINVITE, UAs rollback offer-answer,
   but don't rollback remote target.
   
2. All reliable response to a reINVITE refresh a remote target
   within one INVITE transaction.
   
3. PRACK is used in offer-answer, but is not a target refresh
   request.
   
4. A target-refresh is acknowledged by all SIP request(e.g.
   MESSAGE, INFO, etc.).

An offer-answer and a target-refresh have a different concept
each other. It seems to be unreasonable that those two rules
are applied to both situations at the same time.

Regards,
Shinji

>Another in this category is sending an updated PAI, or changing From. 
>This to is a declaration rather than a negotiation, so that glare 
>shouldn't apply.
>
>In both of those cases, there remains an ordering issue, and perhaps the 
>answer is to forbid behavior that can result in ambiguity. But I fear 
>that cure is worse than the disease, so I'm inclined to seek a better cure.
>
>I'm especially concerned about that in the case of target refresh. If 
>the sender has lost its old address, then we are on the brink of losing 
>the call entirely. Saving the call is more important than whether we 
>have the proper o/a state - that can be fixed after the new target 
>address is established. Also, target refresh cases are typically rare. 
>Two successive ones in close succession in the same direction should be 
>rare^2. (A target refresh of both ends in quick succession could be more 
>likely - for instance if the same event causes both to lose their 
>address. It seems unlikely that this will be a recoverable event.)
>
>So, I think we ought not be forbidding target refresh just because of 
>other state. Ambiguity about which is the old target and which is the 
>new one can be measured by which is a change - assuming there aren't two 
>changes in quick succession. I think Gao has also recommended this.
>
>	Thanks,
>	Paul
>
>>  > > > Look at RFC3311 (especially section 5.2) concerning how to behave
>>  > > > when UPDATE offer is received while there is an outstanding offer.
>>  > >
>>  > > I know this part of 3311. I don't know the meaning of why
>>  > > you highlight this part.
>>  > >
>>  > > Do you think text of our draft has conflict with this part of 3311 here?
>>  > >
>>  > > I don't think so, as our text only for ambiguous cases.
>>  >
>>  > I'm indicating that the RFC3311 mandates the returning 491 or 500
>>  > (depending upon direction).
>>  >
>>  > You are indicating that the RFC3311 text doesn't apply even though
>>  > draft-ietf-sipcore-reinvite-04 sections 3.5 and 4.5 indicate they
>>  > would apply.  RFC3311 section 5.2 includes glare checking as part of
>>  > normal UPDATE processing.
>>  >
>>  > The drafts normative text causing the issue: "When checking for
>>  > glare situations, UAs MUST treat every UPDATE request as if it
>>  > contained an offer."
>>  >
>>  > You keep interpreting/indicating that the normative text does not
>>  > apply to all UPDATEs.  Please provide the normative text which
>>  > indicates your interpretation.
>>  >
>>  > Thanks,
>>  > Brett

From christer.holmberg@ericsson.com  Mon May 31 03:24:35 2010
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
Date: Mon, 31 May 2010 12:24:19 +0200
Thread-Topic: [sipcore] New deal [ was draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP]
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Subject: Re: [sipcore] New deal [ was draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP]
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Hi,=20

>>Below are some questions/ideas on how I think we could=20
>>solve most issues. Some of the stuff shall be seen more as=20
>>brain storming than actual proposal at this point.
>>=20
>>I am fully aware that some of the ideas for sure may limit=20
>>what is said in RFCs etc, so you don't need to tell me about that.
>>=20
>>=20
>>=20
>>=20
>>IDEA #1:  Offer/answer: Usage of SDP o=3D version parameter
>>=20
>>I know this has been discussed at some point, but what is=20
>>the reason we can't use the SDP o=3D version parameter to=20
>>determine in which order the UAS has sent SDPs?
>>=20
>>I believe it would solve offer/answer glare situations=20
>>where the UAC does not known in which order the UAS has sent SDPs.
>>=20
>>That should not be controversial, and does not break any RFC etc.
>=20
>Shinji refuted this one.

Yes, but I think I didn't agree with what he said :)

So, I still think it would work.



=20
>>IDEA #2:   Target update: Only outside re-INVITEs
>>=20
>>Let's forbid target updates "inside" re-INVITEs. I really=20
>>doubt there is a need for those while the UAs are doing=20
>>something else with the re-INVITE.
>>=20
>>That at least solves the glare situation where the remote=20
>>target may be updated in a re-INVITE response and an UPDATE response.
>=20
>For the case of a UA that loses its IP address and must=20
>switch to another, it may be essential to send the UPDATE=20
>immediately. And it won't be getting the response to the=20
>reINVITE in that case, so it can't wait till it does.

Will it EVER get the response to the re-INVITE? As far as I know a target u=
pdate doesn't change the routing of responses. The re-INVITE will most like=
ly time-out. I am not sure the re-INVITE draft talks about re-INVITE timeou=
ts, but I assume it shall be regarded as a failure resposne?

So, eventhough I agree it could happen, I REALLY doubt you are going to be =
able to handle it inside a re-INVITE. As I've said before, in most cases it=
 will also require a new registration.

=20
>In general you should almost never ignore a target refresh,=20
>unless there is something grievous such as an authentication failure.
>=20
>>IDEA #3:    UPDATE: Always use re-INVITE glare rules
>>=20
>>Let's not allow more than one outstanding UPDATE. Ie if a=20
>>UA has sent an UPDATE (with or without SDP), and receives=20
>>another UPDATE request before the answer, it rejects it (same=20
>>rule as for re-INVITE).
>>=20
>>=20
>>That would at least solve the problem where a UA sends an=20
>>UPDATE, receives a 200 OK and a UPDATE request, both with=20
>>different target information.
>=20
>I'm not convinced this solves anything. And if it prevented=20
>an essential target refresh then it would be a bad thing.

The target refresh can still be done in the UPDATE 200 OK.

>As I stated elsewhere, I think glare should be detected=20
>individually for each piece of state being negotiated. Then=20
>the request can be rejected if some form of glare has been detected.

But, if you receive a request first, and a response after, it may be too la=
te to reject the request.

>Target refresh isn't a glare situation, so would never fall into that.

I think the case where you receive UPDATE 200 with target A and UPDATE requ=
est with target B is. In which order were they sent from the remote end? Wh=
ich target is valid?

I just think that, by making some restrictions, we would have a much bigger=
 chance of reaching interoperability, instead of trying to specify complica=
ted rules, which are going to require even more complicated code to impleme=
nt...

Regards,

Christer






> > Ok, let the tomatoes fly...
> >=20
> > ...and let me know if there are any issues which the ideas above do=20
> > NOT solve. I have some more ideas, but I am not sure they=20
> are needed=20
> > so I'll them to myself for now :)
> >=20
> >=20
> > Regards,
> >=20
> > Christer
> >=20
> >=20
> > _______________________________________________
> > sipcore mailing list
> > sipcore@ietf.org
> > https://www.ietf.org/mailman/listinfo/sipcore
> >=20
> =

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your chart:

>> 
>>     A                               B
>>     |                               |
>>   s0|re-INV (offer1)                |s0
>>     |------------------------------>|
>>     |               18x-rel(answer1)|
>>     |<------------------------------|
>>   s1|                        non-2xx|s1
>>     |<------------------------------|<-+
>>   s0|ACK                            |s0|
>>     |----------->                   |  |
>>     |UPD(offer0)                    |  |
>>     |==============================>|* | reject UPDATE
>>     |                       491(UPD)|  |
>>     |<==============================|  |
>>     |                    ACK        |  |
>>     |                    ---------->|<-+
>>     |                               |
>>                 Figure 2



If the UPDATE is in scope of ReINVITE by UAS's view, and the 
modification(*negotiated*) in this UPDATE should not be committed.

If the UPDATE is out scope of ReINVITE by UAS's view, the 
modification(*negotiated*) in this UPDATE can be committed.

As there is a ambiguous for UAS, so UAS should reject it.


> In my recognition, Gao and I are not agreeing.
> 
> I have two comments regarding Figure 2.
> 
> 1. Between a non-2xx to a reINVITE and an ACK, the UAS may
>    accept a receiving UPDATE.
> 
>    This draft seems to forbid the UPDATE.
> 
> 2. In this case, the common view of both UAs has already
>    been synchronized. IMO if an UPDATE has been not used in
>    offer-answer exchange before 4xx, then the original
>    rollback process may work well.
>    That is, a following UPDATE is unnecessary in this case.

And if the UPDATE is in scope of ReINVITE, the state could not be treated 
as synchronized.

Thanks,

Gao


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<br><tt><font size=2>your chart:</font></tt>
<br><tt><font size=2><br>
&gt;&gt; <br>
&gt;&gt; &nbsp; &nbsp; A &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; B<br>
&gt;&gt; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |<br>
&gt;&gt; &nbsp; s0|re-INV (offer1) &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp;|s0<br>
&gt;&gt; &nbsp; &nbsp; |------------------------------&gt;|<br>
&gt;&gt; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
18x-rel(answer1)|<br>
&gt;&gt; &nbsp; &nbsp; |&lt;------------------------------|<br>
&gt;&gt; &nbsp; s1| &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp;non-2xx|s1<br>
&gt;&gt; &nbsp; &nbsp; |&lt;------------------------------|&lt;-+<br>
&gt;&gt; &nbsp; s0|ACK &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;|s0|<br>
&gt;&gt; &nbsp; &nbsp; |-----------&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; | &nbsp;|<br>
&gt;&gt; &nbsp; &nbsp; |UPD(offer0) &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp;| &nbsp;|<br>
&gt;&gt; &nbsp; &nbsp; |==============================&gt;|* | reject UPDATE<br>
&gt;&gt; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; 491(UPD)| &nbsp;|<br>
&gt;&gt; &nbsp; &nbsp; |&lt;==============================| &nbsp;|<br>
&gt;&gt; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp;ACK &nbsp; &nbsp; &nbsp; &nbsp;| &nbsp;|<br>
&gt;&gt; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp;----------&gt;|&lt;-+<br>
&gt;&gt; &nbsp; &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; |<br>
&gt;&gt; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Figure
2</font></tt>
<br>
<br>
<br>
<br><tt><font size=2>If the UPDATE is in scope of ReINVITE by UAS's view,
and the modification(*negotiated*) in this UPDATE should not be committed.</font></tt>
<br>
<br><tt><font size=2>If the UPDATE is out scope of ReINVITE by UAS's view,
the modification(*negotiated*) in this UPDATE can be committed.</font></tt>
<br>
<br><tt><font size=2>As there is a ambiguous for UAS, so UAS should reject
it.</font></tt>
<br>
<br><tt><font size=2><br>
&gt; In my recognition, Gao and I are not agreeing.<br>
&gt; <br>
&gt; I have two comments regarding Figure 2.<br>
&gt; <br>
&gt; 1. Between a non-2xx to a reINVITE and an ACK, the UAS may<br>
&gt; &nbsp; &nbsp;accept a receiving UPDATE.<br>
&gt; <br>
&gt; &nbsp; &nbsp;This draft seems to forbid the UPDATE.<br>
&gt; <br>
&gt; 2. In this case, the common view of both UAs has already<br>
&gt; &nbsp; &nbsp;been synchronized. IMO if an UPDATE has been not used
in<br>
&gt; &nbsp; &nbsp;offer-answer exchange before 4xx, then the original<br>
&gt; &nbsp; &nbsp;rollback process may work well.<br>
&gt; &nbsp; &nbsp;That is, a following UPDATE is unnecessary in this case.</font></tt>
<br>
<br><tt><font size=2>And if the UPDATE is in scope of ReINVITE, the state
could not be treated as synchronized.</font></tt>
<br><tt><font size=2><br>
Thanks,</font></tt>
<br>
<br><tt><font size=2>Gao</font></tt>
<br><br><pre>
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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From gao.yang2@zte.com.cn  Mon May 31 05:50:34 2010
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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> >Target-refresh is somewhat different, because it isn't *negotiated* - 
it 
> >is simply *declared* independently in each direction. So I don't think 
> >it really ought to be considered a glare situation.
> 
> Exactly!
> I've thought the differences between offer-answer and target-
> refresh as below,
> 
> 1. When a UAS sends 4xx to reINVITE, UAs rollback offer-answer,
>    but don't rollback remote target.

If target refresh is committed, it would no be rollback by non-2xx of 
Re-INVITE(not means at any situation of 4xx of Re-INVITE, the remote 
target would be enforced). But it doesn't mean that target refresh or 
other *non-negotiated-declare* is non-rejective. The point here is the 
commitment of target refresh.

And the commitment for target refresh has been defined in the draft, that 
is the first reliable response(a reliable provisional response or a 2xx 
response to the target-refresh request), or a request on the new target 
from UAS. 

[Discussion 1] It seams that this commitment rule can be generalized for 
all *non-negotiated-declare*. And the commitment for target refresh can be 
maintained by two ways: "would no be rollback by non-2xx of Re-INVITE" or 
"restrict UAS send 2xx to Re-INVITE". See details in [Discussion 2].
 
> 2. All reliable response to a reINVITE refresh a remote target
>    within one INVITE transaction.

this is clearly clarfied in section 4.3 of the draft.
(If the UAC receives a reliable provisional response or a 2xx response
   to the target-refresh request, or the UAC receives a request on the
   new target, the UAS has updated its remote target.  The UAC can
   consider the target refresh operation completed.)

> 
> 3. PRACK is used in offer-answer, but is not a target refresh
>    request.

this is 3262's defintion, we need to stick to.

> 
> 4. A target-refresh is acknowledged by all SIP request(e.g.
>    MESSAGE, INFO, etc.).

this is clearly clarfied in section 4.3 of the draft.
(If the UAC receives a reliable provisional response or a 2xx response
   to the target-refresh request, or the UAC receives a request on the
   new target, the UAS has updated its remote target.  The UAC can
   consider the target refresh operation completed.)

[Discussion 2] Current draft has the text in section 4.2 as:

The new
   behavior regarding target-refresh requests implies that a target-
   refresh request can, in some cases, update the remote target even if
   the request is responded with a final error response.  This means
   that target-refresh requests are not atomic.

I think that there's no negative impact on the function and behavior, if 
we restrict the UAS send 2xx to Re-INVITE while there's target refresh(or 
other *non-negotiated-declare*) during it. And if the UAS want to reject 
something *negotiated*, it can send another Re-INVITE-in-scope UPDATE.
If we do so, we can unify the semantic of Re-INVITE atomicity.

Thanks,

Gao


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<br><tt><font size=2><br>
&gt; &gt;Target-refresh is somewhat different, because it isn't *negotiated*
- it <br>
&gt; &gt;is simply *declared* independently in each direction. So I don't
think <br>
&gt; &gt;it really ought to be considered a glare situation.<br>
&gt; <br>
&gt; Exactly!<br>
&gt; I've thought the differences between offer-answer and target-<br>
&gt; refresh as below,<br>
&gt; <br>
&gt; 1. When a UAS sends 4xx to reINVITE, UAs rollback offer-answer,<br>
&gt; &nbsp; &nbsp;but don't rollback remote target.</font></tt>
<br>
<br><tt><font size=2>If target refresh is committed, it would no be rollback
by non-2xx of Re-INVITE(not means at any situation of 4xx of Re-INVITE,
the remote target would be enforced). But it doesn't mean that target refresh
or other *non-negotiated-declare* is non-rejective. The point here is the
commitment of target refresh.</font></tt>
<br>
<br><tt><font size=2>And the commitment for target refresh has been defined
in the draft, that is the first reliable response(a reliable provisional
response or a 2xx response to the target-refresh request), or a request
on the new target from UAS. </font></tt>
<br>
<br><tt><font size=2><b>[Discussion 1]</b> It seams that this commitment
rule can be generalized for all *non-negotiated-declare*. And the commitment
for target refresh can be maintained by two ways: &quot;would no be rollback
by non-2xx of Re-INVITE&quot; or &quot;restrict UAS send 2xx to Re-INVITE&quot;.
See details in [Discussion 2].</font></tt>
<br><tt><font size=2>&nbsp; &nbsp;<br>
&gt; 2. All reliable response to a reINVITE refresh a remote target<br>
&gt; &nbsp; &nbsp;within one INVITE transaction.</font></tt>
<br>
<br><tt><font size=2>this is clearly clarfied in section 4.3 of the draft.</font></tt>
<br><tt><font size=2>(</font></tt><font size=2 face="Courier New">If the
UAC receives </font><font size=2 color=blue face="Courier New">a reliable
provisional response</font><font size=2 face="Courier New"> or a 2xx response</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp;to the target-refresh
request, or the UAC receives a request on the</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp;new target, the UAS has
updated its remote target. &nbsp;The UAC can</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp;consider the target refresh
operation completed.</font><tt><font size=2>)</font></tt>
<br><tt><font size=2><br>
&gt; &nbsp; &nbsp;<br>
&gt; 3. PRACK is used in offer-answer, but is not a target refresh<br>
&gt; &nbsp; &nbsp;request.</font></tt>
<br>
<br><tt><font size=2>this is 3262's defintion, we need to stick to.</font></tt>
<br><tt><font size=2><br>
&gt; &nbsp; &nbsp;<br>
&gt; 4. A target-refresh is acknowledged by all SIP request(e.g.<br>
&gt; &nbsp; &nbsp;MESSAGE, INFO, etc.).</font></tt>
<br>
<br><tt><font size=2>this is clearly clarfied in section 4.3 of the draft.</font></tt>
<br><tt><font size=2>(</font></tt><font size=2 face="Courier New">If the
UAC receives a reliable provisional response or a 2xx response</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp;to the target-refresh
request, </font><font size=2 color=blue face="Courier New">or the UAC receives
a request on the</font>
<br><font size=2 color=blue face="Courier New">&nbsp; &nbsp;new target</font><font size=2 face="Courier New">,
the UAS has updated its remote target. &nbsp;The UAC can</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp;consider the target refresh
operation completed.</font><tt><font size=2>)</font></tt>
<br>
<br><tt><font size=2><b>[Discussion 2]</b> Current draft has the text in
section 4.2 as:</font></tt>
<br>
<br><font size=2 face="Courier New">The new</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp;behavior regarding target-refresh
requests implies that a target-</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp;refresh request can, in
some cases, update the remote target even if</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp;the request is responded
with a final error response. &nbsp;This means</font>
<br><font size=2 face="Courier New">&nbsp; &nbsp;that target-refresh requests
are not atomic.</font>
<br>
<br><tt><font size=2>I think that there's no negative impact on the function
and behavior, if we restrict the UAS send 2xx to Re-INVITE while there's
target refresh(or other *non-negotiated-declare*) during it. And if the
UAS want to reject something *negotiated*, it can send another Re-INVITE-in-scope
UPDATE.</font></tt>
<br><tt><font size=2>If we do so, we can unify the semantic of Re-INVITE
atomicity.</font></tt>
<br>
<br><tt><font size=2>Thanks,</font></tt>
<br>
<br><tt><font size=2>Gao</font></tt>
<br><br><pre>
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From: OKUMURA Shinji <shinji.okumura@softfront.jp>
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATEwithout SDP
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Gao,

>your chart:
>
>>> 
>>>     A                               B
>>>     |                               |
>>>   s0|re-INV (offer1)                |s0
>>>     |------------------------------>|
>>>     |               18x-rel(answer1)|
>>>     |<------------------------------|
>>>   s1|                        non-2xx|s1
>>>     |<------------------------------|<-+
>>>   s0|ACK                            |s0|
>>>     |----------->                   |  |
>>>     |UPD(offer0)                    |  |
>>>     |==============================>|* | reject UPDATE
>>>     |                       491(UPD)|  |
>>>     |<==============================|  |
>>>     |                    ACK        |  |
>>>     |                    ---------->|<-+
>>>     |                               |
>>>                 Figure 2
>
>If the UPDATE is in scope of ReINVITE by UAS's view, and the 
>modification(*negotiated*) in this UPDATE should not be committed.
>
>If the UPDATE is out scope of ReINVITE by UAS's view, the 
>modification(*negotiated*) in this UPDATE can be committed.

You say "the UPDATE is out scope of ReINVITE" or
"the UPDATE is in scope of ReINVITE".
That is meaningless.
The crux of the matter is that a UAS can not recognize
whether the UPDATE is in scope of a ReINVITE or out scope
of it.

>As there is a ambiguous for UAS, so UAS should reject it.

Why?
Even if there is a ambiguous, it's not that a UAS should
reject all ambiguous things. Such a new restriction is
unnecessary.
It is UAC's duty to synchronize a common view.

>> In my recognition, Gao and I are not agreeing.
>> 
>> I have two comments regarding Figure 2.
>> 
>> 1. Between a non-2xx to a reINVITE and an ACK, the UAS may
>>    accept a receiving UPDATE.
>> 
>>    This draft seems to forbid the UPDATE.
>> 
>> 2. In this case, the common view of both UAs has already
>>    been synchronized. IMO if an UPDATE has been not used in
>>    offer-answer exchange before 4xx, then the original
>>    rollback process may work well.
>>    That is, a following UPDATE is unnecessary in this case.
>
>And if the UPDATE is in scope of ReINVITE, the state could not be 
>treated as synchronized.

As I said, an UPDATE is not used. The state is not
ambiguous, the both views are synchronized.

Regards,
Shinji

>Thanks,
>
>Gao

From shinji.okumura@softfront.jp  Mon May 31 07:30:44 2010
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From: OKUMURA Shinji <shinji.okumura@softfront.jp>
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Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATEwithout SDP
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Gao,

>> >Target-refresh is somewhat different, because it isn't *negotiated* - it 
>> >is simply *declared* independently in each direction. So I don't think 
>> >it really ought to be considered a glare situation.
>> 
>> Exactly!
>> I've thought the differences between offer-answer and target-
>> refresh as below,
>> 
>> 1. When a UAS sends 4xx to reINVITE, UAs rollback offer-answer,
>>    but don't rollback remote target.
>
>If target refresh is committed, it would no be rollback by non-2xx of 
>Re-INVITE(not means at any situation of 4xx of Re-INVITE, the remote 
>target would be enforced). But it doesn't mean that target refresh or 
>other *non-negotiated-declare* is non-rejective. The point here is the 
>commitment of target refresh.
>
>And the commitment for target refresh has been defined in the draft, that 
>is the first reliable response(a reliable provisional response or a 2xx 
>response to the target-refresh request), or a request on the new target 
>from UAS. 

Only "first" reliable response?
Is it correct?

Regards,
Shinji

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Cc: "gao.yang2@zte.com.cn" <gao.yang2@zte.com.cn>, "sipcore@ietf.org" <sipcore@ietf.org>, Harbhanu <harbhanu@huawei.com>
Subject: Re: [sipcore] REg. draft-ietf-sipcore-reinvite-04  //Session Timer
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Christer Holmberg wrote:
> Hi, 
> 
>> I agree that there are issues with session-timer.
>>
>> Normally session-timer is simple, because it is negotiated 
>> anew with each exchange. That works reasonably well for 
>> reINVITE and for UPDATE outside of an invite transaction. Its 
>> a mess if an UPDATE within a (re)INVITE transaction 
>> negotiates S-T. I think that needs to be forbidden, which 
>> will be a revision to 4028.
> 
> That could be added to my "NEW DEAL" list of things which shouldn't be done within a re-INVITE :)
> 
>  
>> (A reINVITE should not take so long that a session-timer will 
>> expire before it is complete. Perhaps it should be specified 
>> that *if* a session-timer does expire while a reINVITE is in 
>> progress, that expiration is ignored.)
> 
> I don't think one should try to do too many things with a single re-INVITE (or UPDATE) in the first place.
> 
> If you are going to perform a re-INVITE, which may take some time to finish, do you S-T refresh *before* that, in a separate transaction.

By current definition, you can't do this. Every reinvite and every
update renegotiates the session timer - either a new timer value or else
 the absence of a session timer. So if you want to keep your session
timer going you must respecify it on each reinvite.

It will require a normative change to alter this.

The normative change I advocate is simply the one that says an update
within a reinvite transaction does not affect the session timer. This
however still has issues when there is ambiguity as to whether the
update is within the reinvite or not.

	Thanks,
	Paul

> Regards,
> 
> Christer
> 
> 
> 
> 
> 
> 
>> gao.yang2@zte.com.cn wrote:
>>> Hi,
>>>
>>> Not like O/A usage and target refresh usage, there's no explicit 
>>> definition of allowing subsequent session refresh request 
>> in embedded 
>>> requests(such as UPDATE during pending Re-INVITE). So, I 
>> think it is 
>>> not recommended to sending UPDATE as subsequent session refresh 
>>> request in pending INVITE/Re-INVITE.
>>>
>>> I'd like to talk more about session timer. If people want to do the 
>>> semantics extension for session timer's to allow embedded 
>>> requests(such as UPDATE during pending Re-INVITE), there's one 
>>> suggestion way(INFORMATIVE at all):
>>>
>>> Do it like target refresh, then the 2xx of 
>> INVITE/Re-INVITE's SE value 
>>> should be as the SE value in the last embedded subsequent session 
>>> refresh request's 2xx.
>>>
>>> Then, the glare handling for such usage would be the same as the 
>>> Target Refresh part in draft-ietf-sipcore-reinvite.
>>>
>>> Thanks,
>>>
>>> Gao
>>>
>>> ===================================
>>> Zip    : 210012
>>> Tel    : 87211
>>> Tel2   :(+86)-025-52877211
>>> e_mail : gao.yang2@zte.com.cn
>>> ===================================
>>>
>>>
>>> *Harbhanu <harbhanu@huawei.com>*
>>> ·¢¼þÈË:  sipcore-bounces@ietf.org
>>>
>>> 2010-05-24 13:40
>>>
>>> 	
>>> ÊÕ¼þÈË
>>> 	sipcore@ietf.org
>>> ³­ËÍ
>>> 	
>>> Ö÷Ìâ
>>> 	[sipcore] REg. draft-ietf-sipcore-reinvite-04
>>>
>>>
>>> 	
>>>
>>>
>>>
>>>
>>>
>>>  
>>> 1. UE1 --- INVITE SE-1800 ¨¤ UE2
>>> 2. UE1 <--- 183 ------------------ UE2
>>>  
>>> 3. UE1 --- UPDATE SE-600 ¨¤ UE2
>>> 4. UE1 <--- 200 UPDATE -SE600 -- UE2
>>>  
>>> 5. UE1 <--- 200 INVITE SE-1800 -- UE2
>>>  
>>> Please refer the scenario above for handling of 
>> Session-Timer, (4028).
>>> As per the Draft-ietf-sipcore-reinvite-04 section 4.5:
>>> ¡°  When checking for glare situations, UAs MUST treat every UPDATE
>>>    request as if it contained an offer.  Additionally, UAs 
>> MUST treat
>>>    the exchange of a 2xx response to an INVITE request and its
>>>    corresponding ACK request as an offer/answer exchange.  
>> Consequently,
>>>    the rules regarding glare situations applicable to offer/answer
>>>    exchanges are also applicable to those exchanges.¡±
>>>  
>>> Here, we are not able to conclude whether both UAs should 
>> run session 
>>> timer with 1800 or 600, or otherwise whether the UPDATE should be 
>>> rejected.
>>>  
>>> As per my understanding this draft doesn¡¯t **explicitly** mention 
>>> anything about the Session-Timer, wherein there is an early 
>> negotiation.
>>> Since 4028 also doesn¡¯t
>>> have any reference for early negotiation, so we suggest that it can 
>>> better be incorporated in the above mentioned 
>>> draft[ietf-sipcore-reinvite-04].
>>>  
>>> Please share your opinion and do guide us incase any of the 
>> existing 
>>> RFC/active-draft already captures this.
>>>  
>>> Thanks!
>>>  
>>> Regards,
>>> Harbhanu
>>>  
>>>
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From pkyzivat@cisco.com  Mon May 31 19:18:18 2010
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Cc: OKUMURA Shinji <shinji.okumura@softfront.jp>, sipcore-bounces@ietf.org, "sipcore@ietf.org" <sipcore@ietf.org>, "Christer.Holmberg@ericsson.com" <Christer.Holmberg@ericsson.com>
Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE	without SDP
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gao.yang2@zte.com.cn wrote:

...

> *[Discussion 2]* Current draft has the text in section 4.2 as:
> 
> The new
>    behavior regarding target-refresh requests implies that a target-
>    refresh request can, in some cases, update the remote target even if
>    the request is responded with a final error response.  This means
>    that target-refresh requests are not atomic.
> 
> I think that there's no negative impact on the function and behavior, if 
> we restrict the UAS send 2xx to Re-INVITE while there's target 
> refresh(or other *non-negotiated-declare*) during it. And if the UAS 
> want to reject something *negotiated*, it can send another 
> Re-INVITE-in-scope UPDATE.
> If we do so, we can unify the semantic of Re-INVITE atomicity.

There is always one case when requiring a 2xx response to reinvite is 
totally unacceptable - when it fails authentication. E.g. this could be 
a forged request by a third party. It could even be an attempt by a 
third party to hijack the session by introducing a target change to itself.

There must be other similar cases, such as when the request is malformed.

So IMO requiring a 2xx response is not a successful solution.

	Thanks,
	Paul

From gao.yang2@zte.com.cn  Mon May 31 19:41:38 2010
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Date: Tue, 1 Jun 2010 10:37:53 +0800
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Cc: OKUMURA Shinji <shinji.okumura@softfront.jp>, sipcore-bounces@ietf.org, "sipcore@ietf.org" <sipcore@ietf.org>, "Christer.Holmberg@ericsson.com" <Christer.Holmberg@ericsson.com>
Subject: Re: [sipcore] draft-ietf-sipcore-reinvite-04: glare and UPDATE without SDP
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> > *[Discussion 2]* Current draft has the text in section 4.2 as:
> > 
> > The new
> >    behavior regarding target-refresh requests implies that a target-
> >    refresh request can, in some cases, update the remote target even 
if
> >    the request is responded with a final error response.  This means
> >    that target-refresh requests are not atomic.
> > 
> > I think that there's no negative impact on the function and behavior, 
if 
> > we restrict the UAS send 2xx to Re-INVITE while there's target 
> > refresh(or other *non-negotiated-declare*) during it. And if the UAS 
> > want to reject something *negotiated*, it can send another 
> > Re-INVITE-in-scope UPDATE.
> > If we do so, we can unify the semantic of Re-INVITE atomicity.
> 
> There is always one case when requiring a 2xx response to reinvite is 
> totally unacceptable - when it fails authentication. E.g. this could be 
> a forged request by a third party. It could even be an attempt by a 
> third party to hijack the session by introducing a target change to 
itself.
> 
> There must be other similar cases, such as when the request is 
malformed.
> 
> So IMO requiring a 2xx response is not a successful solution.


Yes. And I considered this case. 

As authentication failure, UAS should reject the target request directly. 
That is, there *MUST NOT* be any 100rel or other in-scope new request (to 
the new target) to do the commitment of the target request.

In other words, as the UAS commit the target request, there *MUST* be 2xx 
of the Re-INVITE.

Thanks,

Gao


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<br><tt><font size=2><br>
&gt; &gt; *[Discussion 2]* Current draft has the text in section 4.2 as:<br>
&gt; &gt; <br>
&gt; &gt; The new<br>
&gt; &gt; &nbsp; &nbsp;behavior regarding target-refresh requests implies
that a target-<br>
&gt; &gt; &nbsp; &nbsp;refresh request can, in some cases, update the remote
target even if<br>
&gt; &gt; &nbsp; &nbsp;the request is responded with a final error response.
&nbsp;This means<br>
&gt; &gt; &nbsp; &nbsp;that target-refresh requests are not atomic.<br>
&gt; &gt; <br>
&gt; &gt; I think that there's no negative impact on the function and behavior,
if <br>
&gt; &gt; we restrict the UAS send 2xx to Re-INVITE while there's target
<br>
&gt; &gt; refresh(or other *non-negotiated-declare*) during it. And if
the UAS <br>
&gt; &gt; want to reject something *negotiated*, it can send another <br>
&gt; &gt; Re-INVITE-in-scope UPDATE.<br>
&gt; &gt; If we do so, we can unify the semantic of Re-INVITE atomicity.<br>
&gt; <br>
&gt; There is always one case when requiring a 2xx response to reinvite
is <br>
&gt; totally unacceptable - when it fails authentication. E.g. this could
be <br>
&gt; a forged request by a third party. It could even be an attempt by
a <br>
&gt; third party to hijack the session by introducing a target change to
itself.</font></tt>
<br><tt><font size=2>&gt; <br>
&gt; There must be other similar cases, such as when the request is malformed.<br>
&gt; <br>
&gt; So IMO requiring a 2xx response is not a successful solution.<br>
</font></tt>
<br>
<br><tt><font size=2>Yes. And I considered this case. </font></tt>
<br>
<br><tt><font size=2>As authentication failure, UAS should reject the target
request directly. That is, there *MUST NOT* be any 100rel or other in-scope
new request (to the new target) to do the commitment of the target request.</font></tt>
<br>
<br><tt><font size=2>In other words, as the UAS commit the target request,
there *MUST* be 2xx of the Re-INVITE.</font></tt>
<br>
<br><tt><font size=2>Thanks,</font></tt>
<br>
<br><tt><font size=2>Gao</font></tt>
<br><br><pre>
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> 
> >> >Target-refresh is somewhat different, because it isn't *negotiated* 
- it 
> >> >is simply *declared* independently in each direction. So I don't 
think 
> >> >it really ought to be considered a glare situation.
> >> 
> >> Exactly!
> >> I've thought the differences between offer-answer and target-
> >> refresh as below,
> >> 
> >> 1. When a UAS sends 4xx to reINVITE, UAs rollback offer-answer,
> >>    but don't rollback remote target.
> >
> >If target refresh is committed, it would no be rollback by non-2xx of 
> >Re-INVITE(not means at any situation of 4xx of Re-INVITE, the remote 
> >target would be enforced). But it doesn't mean that target refresh or 
> >other *non-negotiated-declare* is non-rejective. The point here is the 
> >commitment of target refresh.
> >
> >And the commitment for target refresh has been defined in the draft, 
that 
> >is the first reliable response(a reliable provisional response or a 2xx 

> >response to the target-refresh request), or a request on the new target 

> >from UAS. 
> 
> Only "first" reliable response?
> Is it correct?

This is about the triggering of commitment. After the first reliable 
response or a request to the new target, the target modification is 
committed. 

UAS can send subsequent reliable response with new target.

Thanks,

Gao


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<br><tt><font size=2><br>
&gt; <br>
&gt; &gt;&gt; &gt;Target-refresh is somewhat different, because it isn't
*negotiated* - it <br>
&gt; &gt;&gt; &gt;is simply *declared* independently in each direction.
So I don't think <br>
&gt; &gt;&gt; &gt;it really ought to be considered a glare situation.<br>
&gt; &gt;&gt; <br>
&gt; &gt;&gt; Exactly!<br>
&gt; &gt;&gt; I've thought the differences between offer-answer and target-<br>
&gt; &gt;&gt; refresh as below,<br>
&gt; &gt;&gt; <br>
&gt; &gt;&gt; 1. When a UAS sends 4xx to reINVITE, UAs rollback offer-answer,<br>
&gt; &gt;&gt; &nbsp; &nbsp;but don't rollback remote target.<br>
&gt; &gt;<br>
&gt; &gt;If target refresh is committed, it would no be rollback by non-2xx
of <br>
&gt; &gt;Re-INVITE(not means at any situation of 4xx of Re-INVITE, the
remote <br>
&gt; &gt;target would be enforced). But it doesn't mean that target refresh
or <br>
&gt; &gt;other *non-negotiated-declare* is non-rejective. The point here
is the <br>
&gt; &gt;commitment of target refresh.<br>
&gt; &gt;<br>
&gt; &gt;And the commitment for target refresh has been defined in the
draft, that <br>
&gt; &gt;is the first reliable response(a reliable provisional response
or a 2xx <br>
&gt; &gt;response to the target-refresh request), or a request on the new
target <br>
&gt; &gt;from UAS. <br>
&gt; <br>
&gt; Only &quot;first&quot; reliable response?<br>
&gt; Is it correct?<br>
</font></tt>
<br><tt><font size=2>This is about the triggering of commitment. After
the first reliable response or a request to the new target, the target
modification is committed. </font></tt>
<br>
<br><tt><font size=2>UAS can send subsequent reliable response with new
target.</font></tt>
<br>
<br><tt><font size=2>Thanks,</font></tt>
<br>
<br><tt><font size=2>Gao</font></tt>
<br><br><pre>
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