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Date: Sun, 30 Jun 2002 16:26:50 -0400
From: Henning Schulzrinne <hgs@cs.columbia.edu>
Organization: Columbia University
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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
CC: Sriram Parameswar <sriramp@nortelnetworks.com>, sip@ietf.org
Subject: Re: [Sip] RE: Event Filtering
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Jonathan Rosenberg wrote:
> I acutally like the idea of a more generic filter/trigger mechanism that 
> would apply cross-packages. The tradeoff is one of generality vs. 
> complexity. The single mechanism will be more general, but probably more 
> complex than an event-specific one. How much more complex? Impossible to 
> say, of course. I generally favor generality, so if its not to omuch 
> more complex, I think its a good thing. However, unless you plan on 
> enumerating the set of filters and triggers allowed in a package, I 
> suspect that the general one won't be that much more complex. Saying 
> "only send me notifies once an hour on alternate thursdays, if the value 
> of the foo parameter is bar" is going to be complex in any case.

CPL with the calsch extensions makes expressing alternate Thursdays very 
easy :-)


> 
> I think the Moran draft is a good start, but not quite there yet. Part 
> of the problem is scope. What exactly do we wish to support, in terms of 
> filter/trigger policies? You don't want to boil the ocean with the first 
> version. I think that, if we go for a general solution, we should do 
> JUST enough to meet the *real* problems people are trying to solve, and 
> we can make it extensible to handle more down the road.
> 
> I would postulate that, realistically, the scope of triggers is going to 
> be limited to:
> 
>   * waiting for a certain event to occur within a certain state,
>   * waiting for a certain variable to take a certain value
> 
> Things like the mathematical and temporal conditions are interesting, 
> but I doubt would really be used in practice. Similarly, the filtering 
> conditions are probably limited to asking for notifications to contain 
> only certain variables.
> 
> With a well scoped set of requirements, as I have proposed above, I 
> think you can get a pretty straightforward language that would apply 
> cross-packages.
> 
> I'll also note that, if we assume that all notifications contain XML as 
> bodies (I know its not true, bear with me), then the filter could 
> possibly be an XSLT document. This might be overkill. For things that 
> are not XML, you'd need to specify an XML mapping for it to work (yuck). 
>  Just a thought.
> 
> So, I would propose that, in order to make progress on this issue, we 
> agree upon a realistic scope for these things, in terms of requirements 
> on filters and triggers, and then we can see how complex the 
> general-purpose filtering language is.

We've also looked at extending CPL for presence (draft-wu-cpl-presence, 
I thin). The question is whether the same model makes sense for events.

I don't think XSLT is the right choice. If anything, you want something 
like the XML Query language where you can formulate conditions on XML 
bodies.

Henning






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From daemon@optimus.ietf.org  Mon Jul  1 03:01:42 2002
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One more comment based on further thought and side discussions: While 
one-time URIs are part of the requirements doc, they have a number of 
problems. For example, does this mean that a server can only hand out 
this URI once? What if the client never uses it? How does this interact 
with caching? This adds state complexity and requires transaction-like 
behavior (what if the server crashes just between handing out the 
binding and removing it?).

I would be interested to hear about the problem this feature is supposed 
to solve. One-time URIs are not found elsewhere, from my understanding. 
A web server, say, could always enforce this implicitly, by removing the 
resource addressed by the URI after the first access, but I'm having a 
hard time seeing how this can be done in practice, given that the web 
server won't know how many instances of the AoR will need this information.

James Undery wrote:
> Hi,
> 
> I generally like this draft, although would like to suggest a change. 
> 
> Having "expires=1" a special value isn't very clean, a "usagecount"
> parameter that shows how many times a URL will be valid for seems much
> more flexible. The combination of usagecount and expires would also
> allow for a one-shot URL that's valid for 60 seconds say, i.e. Binding:
> <http://www.example.com/OneTimeToken?12345> ;disposition=token
> ;expires=60 ;usagecount=1
> 
> James
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip




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From daemon@optimus.ietf.org  Mon Jul  1 03:01:44 2002
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Date: Sun, 30 Jun 2002 16:26:50 -0400
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CC: Sriram Parameswar <sriramp@nortelnetworks.com>, sip@ietf.org
Subject: Re: [Sip] RE: Event Filtering
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Jonathan Rosenberg wrote:
> I acutally like the idea of a more generic filter/trigger mechanism that 
> would apply cross-packages. The tradeoff is one of generality vs. 
> complexity. The single mechanism will be more general, but probably more 
> complex than an event-specific one. How much more complex? Impossible to 
> say, of course. I generally favor generality, so if its not to omuch 
> more complex, I think its a good thing. However, unless you plan on 
> enumerating the set of filters and triggers allowed in a package, I 
> suspect that the general one won't be that much more complex. Saying 
> "only send me notifies once an hour on alternate thursdays, if the value 
> of the foo parameter is bar" is going to be complex in any case.

CPL with the calsch extensions makes expressing alternate Thursdays very 
easy :-)


> 
> I think the Moran draft is a good start, but not quite there yet. Part 
> of the problem is scope. What exactly do we wish to support, in terms of 
> filter/trigger policies? You don't want to boil the ocean with the first 
> version. I think that, if we go for a general solution, we should do 
> JUST enough to meet the *real* problems people are trying to solve, and 
> we can make it extensible to handle more down the road.
> 
> I would postulate that, realistically, the scope of triggers is going to 
> be limited to:
> 
>   * waiting for a certain event to occur within a certain state,
>   * waiting for a certain variable to take a certain value
> 
> Things like the mathematical and temporal conditions are interesting, 
> but I doubt would really be used in practice. Similarly, the filtering 
> conditions are probably limited to asking for notifications to contain 
> only certain variables.
> 
> With a well scoped set of requirements, as I have proposed above, I 
> think you can get a pretty straightforward language that would apply 
> cross-packages.
> 
> I'll also note that, if we assume that all notifications contain XML as 
> bodies (I know its not true, bear with me), then the filter could 
> possibly be an XSLT document. This might be overkill. For things that 
> are not XML, you'd need to specify an XML mapping for it to work (yuck). 
>  Just a thought.
> 
> So, I would propose that, in order to make progress on this issue, we 
> agree upon a realistic scope for these things, in terms of requirements 
> on filters and triggers, and then we can see how complex the 
> general-purpose filtering language is.

We've also looked at extending CPL for presence (draft-wu-cpl-presence, 
I thin). The question is whether the same model makes sense for events.

I don't think XSLT is the right choice. If anything, you want something 
like the XML Query language where you can formulate conditions on XML 
bodies.

Henning






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--NextPart

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

	Title		: Session Initiation Protocol Extension for Instant 
                          Messaging
	Author(s)	: B. Campbell, J. Rosenberg
	Filename	: draft-ietf-sip-message-05.txt
	Pages		: 22
	Date		: 28-Jun-02
	
Instant Messageing (IM) refers to the transfer of messages between
users in near real-time.  These messages are usually, but not
required to be, short.  IMs are often used in a conversational mode,
that is, the transfer of messages back and forth is fast enough for
participants to maintain an interactive conversation.
The MESSAGE method is an extension to the Session Initation Protocol
(SIP) that allows the transfer of Instant Messages.  MESSAGE requests
carry the content in the form of MIME body parts.  MESSAGE requests
do not themselves initiate a SIP dialog; under normal usage each
Instant Message stands alone, much like pager messages.  MESSAGE
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other SIP request.
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From daemon@optimus.ietf.org  Mon Jul  1 07:58:30 2002
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From: "Alexeitsev, D" <D.Alexeitsev@telekom.de>
To: christer.holmberg@lmf.ericsson.se
Cc: sip@ietf.org, chogg@nortelnetworks.com
Subject: Re: [Sip] Local vs Remote ringback on 180
Date: Mon, 1 Jul 2002 13:17:13 +0200 
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Hello All,

Some comments/questions inline

Greetings,
Denis Alexeitsev

-----Ursprüngliche Nachricht-----
Von: Christer Holmberg [mailto:christer.holmberg@lmf.ericsson.se]
Gesendet am: Freitag, 28. Juni 2002 18:53
An: Chris Hogg
Cc: Flemming Andreasen; Jonathan Rosenberg; Mark Watson; sip@ietf.org;
Bob Penfield
Betreff: Re: [Sip] Local vs Remote ringback on 180

[CHH] The UAS may still send early media, but since it sends 180 without
SDP it allows the UAC to generate local ringback tone, so the UAC now
can assume that it can "cut" the early media from the caller during that
time. If the UAS does NOT want the UAC to generate local ringback, the
180 would also contain SDP (the same SDP which was used in the 183 to
establish the early dialog, ie the SDP is not a new offer/answer -
simply a ringback "indicator").

[DA]I have a question about the need to receive the SDP in any responses in order to play remote ringback. As the offer/answer draft says:
"Once the offerer has sent the offer, it MUST be prepared to receive media"

The SDP coming from the other side is used to send the media, that is unimportant in the ringback case. Am I missing something here?

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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Date: Mon, 01 Jul 2002 14:26:09 +0300
From: Christer Holmberg <christer.holmberg@lmf.ericsson.se>
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To: "Alexeitsev, D" <D.Alexeitsev@telekom.de>
CC: sip@ietf.org, chogg@nortelnetworks.com
Subject: Re: [Sip] Local vs Remote ringback on 180
References: <A89A213731F7D51196A0000347055C83E392AC@G9JNT.mgb01.telekom.de>
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Hi,

"Alexeitsev, D" wrote:

> Hello All,
>
> Some comments/questions inline
>
> Greetings,
> Denis Alexeitsev
>
> -----Ursprüngliche Nachricht-----
> Von: Christer Holmberg [mailto:christer.holmberg@lmf.ericsson.se]
> Gesendet am: Freitag, 28. Juni 2002 18:53
> An: Chris Hogg
> Cc: Flemming Andreasen; Jonathan Rosenberg; Mark Watson; sip@ietf.org;
> Bob Penfield
> Betreff: Re: [Sip] Local vs Remote ringback on 180
>
> [CHH] The UAS may still send early media, but since it sends 180 without
> SDP it allows the UAC to generate local ringback tone, so the UAC now
> can assume that it can "cut" the early media from the caller during that
> time. If the UAS does NOT want the UAC to generate local ringback, the
> 180 would also contain SDP (the same SDP which was used in the 183 to
> establish the early dialog, ie the SDP is not a new offer/answer -
> simply a ringback "indicator").
>
> [DA]I have a question about the need to receive the SDP in any responses in order to play remote ringback. As the offer/answer draft says:
> "Once the offerer has sent the offer, it MUST be prepared to receive media"
>
> The SDP coming from the other side is used to send the media, that is unimportant in the ringback case. Am I missing something here?

[CHH] For the specific ringback tone case one can argue that an SDP is not needed for the media itself (unless you need it for RTCP purpose,
or something).

However, there is no specific "remote ringback session" in SIP. You establish a generic early dialiog, by sending an offer and receiving an
answer, which can then be used for any kind of early media (ringback tone, announcement,...), in any direction (depending on the "direction"
indicators - sendrecv, sendonly, inactive etc etc).

Regards,

Christer Holmberg
Ericsson Finland



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Mon Jul  1 10:05:49 2002
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From: "Mark Watson" <mwatson@nortelnetworks.com>
To: "'Christer Holmberg'" <christer.holmberg@lmf.ericsson.se>,
        "Alexeitsev, D" <D.Alexeitsev@telekom.de>
Cc: sip@ietf.org, "Chris Hogg" <chogg@nortelnetworks.com>
Subject: RE: [Sip] Local vs Remote ringback on 180
Date: Mon, 1 Jul 2002 14:35:04 +0100 
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All,

I'm confused by this discussion, because there seems to be a desire to
extrapolate some semantics about the *media* from a message which does =
not
contain SDP. Apologies if I have missed the issue at hand, or if the
following is obvious, but...

The unify model means that media and session state are completely
independent, so, to talk about '180 with SDP' or '180 without SDP' =
makes no
sense.

You can talk about the effect 180 has on the session state (indicates =
that
called party is being alerted), but this has nothing to do with the =
media
state (if there was media before, there still is, if there was no media
before, there still isn't).

You can talk about the effect that SDP offer/answer exchanges have on =
the
media, but this operates independently of the SIP message those SDP =
bodies
arrive in.

So,

1) the UAC knows whether there is media or not (based on SDP =
offer/answer
exchanges so far)
2) if 180 has been received, the UAC knows that the called party is =
alerting

If the UAC knows that the called party is alerting AND knows that there =
is
media, it is a local matter what to do. Personally, I cannot see an =
argument
for doing anything other than playing the media (it might contain =
'ringing
with call waiting announcement').

A PROBLEM arises if there is a UAS which negotiates media but then =
sends
silence (no packets) - there ARE some reasons why it might want to do =
this
to do with reducing clipping on answer. The result is that silence =
would be
all the caller would hear even though the UAC may know that the called =
party
is being alerted. I think we just have to say that it is foolish for =
the UAS
to negotiate media and then send silence - it should put the media on =
hold
if it has nothing to send.

Regards,

Mark

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@lmf.ericsson.se]
> Sent: 01 July 2002 12:26
> To: Alexeitsev, D
> Cc: sip@ietf.org; Hogg, Chris [MOP:MA01:EXCH]
> Subject: Re: [Sip] Local vs Remote ringback on 180
>=20
>=20
>=20
> Hi,
>=20
> "Alexeitsev, D" wrote:
>=20
> > Hello All,
> >
> > Some comments/questions inline
> >
> > Greetings,
> > Denis Alexeitsev
> >
> > -----Urspr=FCngliche Nachricht-----
> > Von: Christer Holmberg [mailto:christer.holmberg@lmf.ericsson.se]
> > Gesendet am: Freitag, 28. Juni 2002 18:53
> > An: Chris Hogg
> > Cc: Flemming Andreasen; Jonathan Rosenberg; Mark Watson;=20
> sip@ietf.org;
> > Bob Penfield
> > Betreff: Re: [Sip] Local vs Remote ringback on 180
> >
> > [CHH] The UAS may still send early media, but since it=20
> sends 180 without
> > SDP it allows the UAC to generate local ringback tone, so=20
> the UAC now
> > can assume that it can "cut" the early media from the=20
> caller during that
> > time. If the UAS does NOT want the UAC to generate local=20
> ringback, the
> > 180 would also contain SDP (the same SDP which was used in=20
> the 183 to
> > establish the early dialog, ie the SDP is not a new offer/answer -
> > simply a ringback "indicator").
> >
> > [DA]I have a question about the need to receive the SDP in=20
> any responses in order to play remote ringback. As the=20
> offer/answer draft says:
> > "Once the offerer has sent the offer, it MUST be prepared=20
> to receive media"
> >
> > The SDP coming from the other side is used to send the=20
> media, that is unimportant in the ringback case. Am I missing=20
> something here?
>=20
> [CHH] For the specific ringback tone case one can argue that=20
> an SDP is not needed for the media itself (unless you need it=20
> for RTCP purpose,
> or something).
>=20
> However, there is no specific "remote ringback session" in=20
> SIP. You establish a generic early dialiog, by sending an=20
> offer and receiving an
> answer, which can then be used for any kind of early media=20
> (ringback tone, announcement,...), in any direction=20
> (depending on the "direction"
> indicators - sendrecv, sendonly, inactive etc etc).
>=20
> Regards,
>=20
> Christer Holmberg
> Ericsson Finland
>=20
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

------_=_NextPart_001_01C22104.1BBE12E4
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charset=3DISO-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2655.35">
<TITLE>RE: [Sip] Local vs Remote ringback on 180</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>All,</FONT>
</P>

<P><FONT SIZE=3D2>I'm confused by this discussion, because there seems =
to be a desire to extrapolate some semantics about the *media* from a =
message which does not contain SDP. Apologies if I have missed the =
issue at hand, or if the following is obvious, but...</FONT></P>

<P><FONT SIZE=3D2>The unify model means that media and session state =
are completely independent, so, to talk about '180 with SDP' or '180 =
without SDP' makes no sense.</FONT></P>

<P><FONT SIZE=3D2>You can talk about the effect 180 has on the session =
state (indicates that called party is being alerted), but this has =
nothing to do with the media state (if there was media before, there =
still is, if there was no media before, there still isn't).</FONT></P>

<P><FONT SIZE=3D2>You can talk about the effect that SDP offer/answer =
exchanges have on the media, but this operates independently of the SIP =
message those SDP bodies arrive in.</FONT></P>

<P><FONT SIZE=3D2>So,</FONT>
</P>

<P><FONT SIZE=3D2>1) the UAC knows whether there is media or not (based =
on SDP offer/answer exchanges so far)</FONT>
<BR><FONT SIZE=3D2>2) if 180 has been received, the UAC knows that the =
called party is alerting</FONT>
</P>

<P><FONT SIZE=3D2>If the UAC knows that the called party is alerting =
AND knows that there is media, it is a local matter what to do. =
Personally, I cannot see an argument for doing anything other than =
playing the media (it might contain 'ringing with call waiting =
announcement').</FONT></P>

<P><FONT SIZE=3D2>A PROBLEM arises if there is a UAS which negotiates =
media but then sends silence (no packets) - there ARE some reasons why =
it might want to do this to do with reducing clipping on answer. The =
result is that silence would be all the caller would hear even though =
the UAC may know that the called party is being alerted. I think we =
just have to say that it is foolish for the UAS to negotiate media and =
then send silence - it should put the media on hold if it has nothing =
to send.</FONT></P>

<P><FONT SIZE=3D2>Regards,</FONT>
</P>

<P><FONT SIZE=3D2>Mark</FONT>
</P>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Christer Holmberg [<A =
HREF=3D"mailto:christer.holmberg@lmf.ericsson.se">mailto:christer.holmbe=
rg@lmf.ericsson.se</A>]</FONT>
<BR><FONT SIZE=3D2>&gt; Sent: 01 July 2002 12:26</FONT>
<BR><FONT SIZE=3D2>&gt; To: Alexeitsev, D</FONT>
<BR><FONT SIZE=3D2>&gt; Cc: sip@ietf.org; Hogg, Chris =
[MOP:MA01:EXCH]</FONT>
<BR><FONT SIZE=3D2>&gt; Subject: Re: [Sip] Local vs Remote ringback on =
180</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Hi,</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &quot;Alexeitsev, D&quot; wrote:</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Hello All,</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Some comments/questions inline</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Greetings,</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Denis Alexeitsev</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; -----Urspr=FCngliche Nachricht-----</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Von: Christer Holmberg [<A =
HREF=3D"mailto:christer.holmberg@lmf.ericsson.se">mailto:christer.holmbe=
rg@lmf.ericsson.se</A>]</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Gesendet am: Freitag, 28. Juni 2002 =
18:53</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; An: Chris Hogg</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Cc: Flemming Andreasen; Jonathan =
Rosenberg; Mark Watson; </FONT>
<BR><FONT SIZE=3D2>&gt; sip@ietf.org;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Bob Penfield</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; Betreff: Re: [Sip] Local vs Remote =
ringback on 180</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; [CHH] The UAS may still send early media, =
but since it </FONT>
<BR><FONT SIZE=3D2>&gt; sends 180 without</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; SDP it allows the UAC to generate local =
ringback tone, so </FONT>
<BR><FONT SIZE=3D2>&gt; the UAC now</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; can assume that it can &quot;cut&quot; the =
early media from the </FONT>
<BR><FONT SIZE=3D2>&gt; caller during that</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; time. If the UAS does NOT want the UAC to =
generate local </FONT>
<BR><FONT SIZE=3D2>&gt; ringback, the</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; 180 would also contain SDP (the same SDP =
which was used in </FONT>
<BR><FONT SIZE=3D2>&gt; the 183 to</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; establish the early dialog, ie the SDP is =
not a new offer/answer -</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; simply a ringback =
&quot;indicator&quot;).</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; [DA]I have a question about the need to =
receive the SDP in </FONT>
<BR><FONT SIZE=3D2>&gt; any responses in order to play remote ringback. =
As the </FONT>
<BR><FONT SIZE=3D2>&gt; offer/answer draft says:</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; &quot;Once the offerer has sent the offer, =
it MUST be prepared </FONT>
<BR><FONT SIZE=3D2>&gt; to receive media&quot;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt;</FONT>
<BR><FONT SIZE=3D2>&gt; &gt; The SDP coming from the other side is used =
to send the </FONT>
<BR><FONT SIZE=3D2>&gt; media, that is unimportant in the ringback =
case. Am I missing </FONT>
<BR><FONT SIZE=3D2>&gt; something here?</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; [CHH] For the specific ringback tone case one =
can argue that </FONT>
<BR><FONT SIZE=3D2>&gt; an SDP is not needed for the media itself =
(unless you need it </FONT>
<BR><FONT SIZE=3D2>&gt; for RTCP purpose,</FONT>
<BR><FONT SIZE=3D2>&gt; or something).</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; However, there is no specific &quot;remote =
ringback session&quot; in </FONT>
<BR><FONT SIZE=3D2>&gt; SIP. You establish a generic early dialiog, by =
sending an </FONT>
<BR><FONT SIZE=3D2>&gt; offer and receiving an</FONT>
<BR><FONT SIZE=3D2>&gt; answer, which can then be used for any kind of =
early media </FONT>
<BR><FONT SIZE=3D2>&gt; (ringback tone, announcement,...), in any =
direction </FONT>
<BR><FONT SIZE=3D2>&gt; (depending on the &quot;direction&quot;</FONT>
<BR><FONT SIZE=3D2>&gt; indicators - sendrecv, sendonly, inactive etc =
etc).</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Regards,</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; Christer Holmberg</FONT>
<BR><FONT SIZE=3D2>&gt; Ericsson Finland</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>&gt; This list is for NEW development of the core =
SIP Protocol</FONT>
<BR><FONT SIZE=3D2>&gt; Use sip-implementors@cs.columbia.edu for =
questions on current sip</FONT>
<BR><FONT SIZE=3D2>&gt; Use sipping@ietf.org for new developments on =
the application of sip</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

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Use sip-implementors@cs.columbia.edu for questions on current sip
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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
CC: Sean Olson <seancolson@yahoo.com>, Cullen Jennings <fluffy@cisco.com>,
        sip@ietf.org
Subject: Re: [Sip] refreshing NAT binding for SIP with UDP registrations
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Jonathan Rosenberg wrote:
> 
> The holes that this "CRLF approach" has are similar to some of the
> generic problems of STUN, which I would prefer be kept isolated there.
> THe main one really is that the frequency of the refreshes is dependent
> on the binding lifetime; that is not generally known, either to the UA,
> or the registrar. TCP works better there. Because TCP connections have
> an explicit teardown, they are held in NATs for a long time without
> keepalives.
> 
> There is also the needless network traffic from these 1 byte UDP packets
> flying around (with 8 bytes UDP, 20 bytes IPv4 headers), a huge waste.

I agree on both points...and the majority of corporate network admins
would also agree with this, not to mention ISP's if they continue to see
constant keepalives....as opposed to what appears to be valid traffic,
they may be inclined to cancel service. 

SIP clients could be considered a violation of ISP policy (in the sense
that the majority of existing ISP's forbid web servers on home accounts,
for instance, at this point the client represents an internal service
listening for connections), excess traffic would cause more notice to
these services. 

I only mention this since many of the NAT scenarios described in past
drafts offering solutions for SIP, that I have read, seem to be more
suited to SOHO NAT solutions rather than intra/extranet entities, and
many of the solutions would not be considered as valid solutions by many
corporate administrators/policies due to security issues that would
arise from the solutions operation. 

To clarify the latter comment SIP traversal solutions for the enterprise
that involves keep alives or topology discovery on enterprise firewalls
may be considered a violation of enterprise security policy, not to
mention the excess traffic, which is typically hunted down to be
silenced on many of these types of networks, and lastly the potential
risk of providing additional vulnerabilities to the internal network
that are yet unknown. Many infrastructures do not want internal users
having knowlege of the topology. Placing the ability to do so on a
client to determine this information may be frowned upon as well.

Anyway the firewall vendors have begun to provide SIP solutions to
address this now, so it is becoming a moot point. MIDCOM may provide a
way to offer additional support for many items other than firewalls,
such as load balancing devices etc, so this has much merit, but a
traversal solution, IMHO, is not necessary or acceptable, except in the
SOHO realm, if that. 

But if we must work on and develop this capability then the direction
that you stated above is the way I would go as well because there are
other implications and benefits that we may possibly get from doing this
in addition to reducing the excess traffic.

Chris Martin

> 
> -Jonathan R.
> 
> Sean Olson wrote:
> > This seems to have too many holes.
> > However, it *might* be useful to define a
> > lightweight SIP keep-alive mechanism.
> >
> > /sean
> >
> > --- Cullen Jennings <fluffy@cisco.com> wrote:
> >
> >>One point about UDP registrations, I don't know if
> >>you really need to send a
> >>true registration, you can just send any message
> >>that creates minimal load
> >>on the proxy. For example the UA could send a single
> >>CRLF to the proxy. The
> >>proxy could have some code to silently ignore any
> >>message that is clearly
> >>too short to be a valid SIP message. Obviously this
> >>won't work in the cases
> >>where the UA sends to a proxy inside the NAT then
> >>the proxy sends to a
> >>another proxy outside the NAT. It also has the
> >>problems with no response so
> >>you don't know if the message made it to the NAT or
> >>not.
> >>
> >>Do folks see this as an ideal worth pursing or a
> >>really bad idea that we
> >>should suggest people don't do?
> >>
> >>Cullen
> >>
> >>
> >>_______________________________________________
> >>Sip mailing list
> >>https://www1.ietf.org/mailman/listinfo/sip
> >>This list is for NEW development of the core SIP
> >>Protocol
> >>Use sip-implementors@cs.columbia.edu for questions
> >>on current sip
> >>Use sipping@ietf.org for new developments on the
> >>application of sip
> >
> >
> >
> > __________________________________________________
> > Do You Yahoo!?
> > Yahoo! - Official partner of 2002 FIFA World Cup
> > http://fifaworldcup.yahoo.com
> >
> 
> --
> Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
> Chief Scientist                         First Floor
> dynamicsoft                             East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
> http://www.jdrosen.net                  PH:  (973) 952-5000
> http://www.dynamicsoft.com
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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From daemon@optimus.ietf.org  Mon Jul  1 10:21:18 2002
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From: "Salva Rey" <salreyca@hotmail.com>
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Hi all,

a doubt regarding forking proxies,

why would a proxy do if forked an INVITE request, it got back some reliable 
provisional responses that established some early dialogs at the UAC, and 
then one of the UAS's sends a final response 200 oK. If the other early 
dialogs try to send further reliable provisional responses this wont get 
through the proxy since a final response was already sent on the response 
context. Should the proxy behaviour not only forward immediately 200ok 
responses to INVITE, but also reliable responses ?

best regards
Salva

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From daemon@optimus.ietf.org  Mon Jul  1 11:21:46 2002
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CC: "Alexeitsev, D" <D.Alexeitsev@telekom.de>, sip@ietf.org,
        Chris Hogg <chogg@nortelnetworks.com>
Subject: Re: [Sip] Local vs Remote ringback on 180
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Hi Mark,

> I'm confused by this discussion, because there seems to be a desire to
> extrapolate some semantics about the *media* from a message which does
> not contain SDP. Apologies if I have missed the issue at hand, or if
> the following is obvious, but...
>
> The unify model means that media and session state are completely
> independent, so, to talk about '180 with SDP' or '180 without SDP'
> makes no sense.

I agree, but those are the terms used by the drafts I've refered to. The
ISUP/SIP mapping draft says that "180 with SDP" means that a remote
ringback will be generated, and bis-09 says that "180 without SDP" MAY
cause the UAC to generate a local ringback.

Of course, those statements may sometimes be valid, if we receive either
a 180 without SDP, OR we receive 180 with SDP - and no early dialog
still exists. My orignial question was what we do if we receive 180
without SDP, but we DO have an early dialog established. It has been
argued that the UAC should not generate local ringback in that case -
instead assume the UAS does it since there is an early dialog
established (if the UAS wants the UAC to generate local ringback it
would have to put the media on hold using UPDATE). IF that's the case, I
think it should me more clarified. Clarification is the only thing I've
asked for - no changes in the protocol.

> You can talk about the effect 180 has on the session state (indicates
> that called party is being alerted), but this has nothing to do with
> the media state (if there was media before, there still is, if there
> was no media before, there still isn't).

I've said all the time that the session state is not affected, only that
the 180 affects what the application does with the media.

So, what you just wrote means, that in the case we have an established
early dialog it really doesn't matter, from a ringback point of view, if
there is a SDP in the 180 or not. Is that the "official" clarification
I've asked for? :)

> You can talk about the effect that SDP offer/answer exchanges have on
> the media, but this operates independently of the SIP message those
> SDP bodies arrive in.

True. That is why I also asked if the SDP is a good "ringback indicator"
in the first place...

> So,
>
> 1) the UAC knows whether there is media or not (based on SDP
> offer/answer exchanges so far)
> 2) if 180 has been received, the UAC knows that the called party is
> alerting
>
> If the UAC knows that the called party is alerting AND knows that
> there is media, it is a local matter what to do. Personally, I cannot
> see an argument for doing anything other than playing the media (it
> might contain 'ringing with call waiting announcement').

I think it was earlier said that the decission can't be based on if
there is media or not. There may be RTP packets, but that doesn't mean
they contain any "real" media.

> A PROBLEM arises if there is a UAS which negotiates media but then
> sends silence (no packets) - there ARE some reasons why it might want
> to do this to do with reducing clipping on answer. The result is that
> silence would be all the caller would hear even though the UAC may
> know that the called party is being alerted. I think we just have to
> say that it is foolish for the UAS to negotiate media and then send
> silence - it should put the media on hold if it has nothing to send.

Like I said earlier, that would probably not happend when the same
server establishes the early dialog and sends the 180.  I earlier gave
an example where a B2BUA sends the announcement ("you have now reached
the X company"), the contacts a SIP phone which sends 180 (the SIP phone
does NOT generate early media). Now, the B2BUA gets the 180, stop
playing the announcement, and forwards the 180 to the UAC. Now, since
there are no more announcements, and the SIP phone doesn't generate
remote ringback, and the UAC doesn't generate local ringback, there is
no ringback at all. In this case the B2BUA would have to send UPDATE
(media on hold), then forward the 180 (the UAC will now generate local
ringback), and when the SIP phone answers it would have to send another
UPDATE (media un-hold) and then forward the 200.

An issue related to this I brought up earlier, is if the UAS will have
to buffer the 200 until both UPDATE transactions have been completed.
Any comments on that?

Also, in a non-PRACK environment, is a 18x "allowed" to establish an
early dialog in the first place? Bis-09 says that the answer must be
sent in a reliable response, which is 200 OK. Yes, bis-09 DOES say that
the answer MAY also be sent earlier, unreliably in a 18x, but when is
the early dialog actually "officially" established? Again, the UAC could
make a ringback decission based on if there is a SDP in the 180 or not,
but a unreliably sent 180 may not even arrive...


Regards,

Christer Holmberg
Ericsson Finland



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From: "Alexeitsev, D" <D.Alexeitsev@telekom.de>
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Subject: Re: [Sip] Local vs Remote ringback on 180
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Hello

a comment inline

Greeting,
Denis Alexeitsev

> -----Ursprüngliche Nachricht-----
> Von: Christer Holmberg [mailto:christer.holmberg@lmf.ericsson.se]
> Gesendet am: Freitag, 28. Juni 2002 18:53
> An: Chris Hogg
> Cc: Flemming Andreasen; Jonathan Rosenberg; Mark Watson; sip@ietf.org;
> Bob Penfield
> Betreff: Re: [Sip] Local vs Remote ringback on 180


However, there is no specific "remote ringback session" in SIP. You establish a generic early dialiog, by sending an offer and receiving an answer,

[DA]As the offer/answer draft says: one "MUST be prepared to receive media" after the sending of the offer, so it does not require for UAC to receive an answer in order to start the media playback (as the early media, in my understanding, always come from the UAS i.e. ringback tone, announcement,... and not other way around)





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Subject: [Sip] 481 response by a proxy ?
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1)

The SIP RFC2543bis-09 only mentions 481 (call leg/transaction does not =
exist) as a response generated by a UAS. I donot see any specific =
mention of this non-2xx response generated by a proxy which had done =
record-route for receiving mid-dialog requests like re-INVITE.


2)
The SIP RFC2543bis-09 does not seem to discuss the behaviour of a dialog =
stateful proxy. Does it mean, the handling of mid-dialog requests like =
re-INVITEs is left to the implementation, in case there is no match =
found on the dialog context ? Or should the dialog stateful proxy =
behaviour be same as a UAS, in which case it should generate a 481 =
response ?

3)

What is the behaviour of a transaction stateful proxy (no dialog states =
maintained) when it receives a re-INVITE on a secondary server =
(identified by DNS) because of a primary server reboot. Does it create a =
new client transaction, forward the re-INVITE to the next hop ? Does it =
behave statelessly ? The RFC seems to be clear on handling a response =
recieved at a transaction stateful proxy (which does not contain the =
response context) by saying that the proxy MUST forward the response by =
behaving as a stateless proxy.
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</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>1)</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>The SIP RFC2543bis-09 only mentions 481 =
(call=20
leg/transaction does not exist) as a response generated by a UAS. I =
donot see=20
any specific mention of this non-2xx response generated by a proxy which =
had=20
done record-route for receiving mid-dialog requests like =
re-INVITE.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2></FONT><FONT face=3DArial =
size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>2)</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>The SIP RFC2543bis-09 does not seem to =
discuss the=20
behaviour of a dialog stateful proxy. Does it mean, the handling of =
mid-dialog=20
requests like re-INVITEs is left to the implementation, in case there is =
no=20
match found on the dialog context ? Or should the dialog stateful proxy=20
behaviour be same as a UAS, in which case it should generate a 481 =
response=20
?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV>
<DIV><FONT face=3DArial size=3D2>3)</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>What is the behaviour of a transaction =
stateful=20
proxy (no dialog states maintained) when it receives a re-INVITE on a =
secondary=20
server (identified by DNS) because of a primary server reboot. Does it =
create a=20
new client transaction, forward the re-INVITE to the next hop ? Does it =
behave=20
statelessly ? The RFC seems to be clear on handling a response recieved =
at a=20
transaction stateful proxy (which does not contain the response context) =
by=20
saying that the proxy MUST forward the response by behaving as a =
stateless=20
proxy.</FONT></DIV></DIV></BODY></HTML>

------=_NextPart_000_0379_01C2212D.489AE1B0--


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From daemon@optimus.ietf.org  Mon Jul  1 14:22:18 2002
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To: "'Alexeitsev, D'" <D.Alexeitsev@telekom.de>,
        <christer.holmberg@lmf.ericsson.se>
Cc: <sip@ietf.org>
Subject: RE: [Sip] Local vs Remote ringback on 180
Date: Mon, 1 Jul 2002 13:30:22 -0400
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comments inline...

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of
> Alexeitsev, D
> Sent: Monday, July 01, 2002 11:30 AM
> To: christer.holmberg@lmf.ericsson.se
> Cc: sip@ietf.org
> Subject: Re: [Sip] Local vs Remote ringback on 180
>
>
> Hello
>
> a comment inline
>
> Greeting,
> Denis Alexeitsev
>
> > -----Ursprüngliche Nachricht-----
> > Von: Christer Holmberg [mailto:christer.holmberg@lmf.ericsson.se]
> > Gesendet am: Freitag, 28. Juni 2002 18:53
> > An: Chris Hogg
> > Cc: Flemming Andreasen; Jonathan Rosenberg; Mark Watson;
> sip@ietf.org;
> > Bob Penfield
> > Betreff: Re: [Sip] Local vs Remote ringback on 180
>
>
> However, there is no specific "remote ringback session" in
> SIP. You establish a generic early dialiog, by sending an
> offer and receiving an answer,
>
> [DA]As the offer/answer draft says: one "MUST be prepared to
> receive media" after the sending of the offer, so it does not
> require for UAC to receive an answer in order to start the
> media playback (as the early media, in my understanding,
> always come from the UAS i.e. ringback tone, announcement,...
> and not other way around)
>

We want the UAC to render received media once it's sent an offer to
prevent clipping.  We need this, it causes an unacceptable user
experience if we don't do this.  In addition, we don't want to require
the UAS to provide "call progress" media, we have the 180 for it to
indicate the callee is being alerted, the UAC can convey this state to
the caller through local ringback or whatever.  It's been
proposed/discussed/and I believe there's consensus that a UAS will send
183 (or 180 recently) with SDP when it intends to present media that
should be rendered to the caller.  (Now this is not required of the UAS
but it is "polite" in that it informs the UAC of its intentions.)  The
expected UAC behavior is for it to stop "ringback" indication at this
point.

Now there's the case when the UAS is doing something more with the
session, and stops sending media to the UAC.  In this case, the caller
will not see/hear any "call progress".  What's being asked for now is
clarification on how to "ask" the UAC to again generate call progress
indication.  It's been suggested that another 18x without SDP indicate
this scenario.  As Christer indicated, there are some I-Ds that describe
this behavior.  There was a period in time where there seemed to be
consensus in the WG on this.  However, offer-answer, unify, ... I-Ds
have since come along that more rigorously define "early dialogs",
meaning of SDP, etc.  And again, the presence of SDP in 18x has nothing
to do with when a UAC should generate or expect session progress
indications.  AFAIK, it's not possible in SIP to indicate to a UAC that
it should stop rendering media and go back to local generation of
progress indication.  Nor do we want to try to do this in SIP.

IMHO, services/applications that will generate early media and then stop
have the responsibility to make the caller's user experience acceptable
(i.e., by playing an announcement, music, or whatever).  In Christer's
B2BUA example, the B2BUA's UAC assumes the responsibility to provide
call progress indication to its user (which is another UAC).

We should only recommend that a UAC generate a local indication upon
receipt of a 180 and then stop upon receipt of media (which it should
render).  Services such as Christer's example must take it upon
themselves to provide an acceptable user experience.  Otherwise we add
complexity to the protocol and degrade both interoperability and user
experience.

This is what I'd propose be the "official" clarification.  My 2 cents
anyway.

Regards,
Bert

>
>
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


_______________________________________________
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From daemon@optimus.ietf.org  Mon Jul  1 15:44:20 2002
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From: "Pete Cordell" <pcordell@ridgewaysystems.com>
To: "Cullen Jennings" <fluffy@cisco.com>, <sip@ietf.org>
Cc: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
References: <DLEHICEBMNEIPCACNLPCOEOLCDAA.fluffy@cisco.com>
Subject: Re: [Sip] Proposed change in draft-ietf-sip-nat
Date: Mon, 1 Jul 2002 18:24:28 +0100
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Simply because it is the agreed strategy of the pre-midcom design group to
define something called SPAN.

It may not look like SPAN-A, and it may look more like TURN, but there is a
way to go before that happens.

Much of the mentioned the lack of consensus has been down to a lack of
reading rather than any identified technical issues, so I suggest we go with
the agreed name rather than starting a bun fight before we've identified and
discussed any technical problems.  To do otherwise is just creating ill will
at a time we don't need it (at least I'm assuming that people don't want it
at this stage).

Pete.


----- Original Message -----
From: "Cullen Jennings" <fluffy@cisco.com>
To: "Pete Cordell" <pcordell@ridgewaysystems.com>; <sip@ietf.org>
Cc: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
Sent: 28 June 2002 19:35
Subject: RE: [Sip] Proposed change in draft-ietf-sip-nat


>
> My read of the SPAN document was it did not even represent consensus of
the
> design team much less the midcom WG.
>
> Why should references to TURN or SPAM be changed to just SPAN?
>
> Thanks, Cullen
>
> > -----Original Message-----
> > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Pete
> > Cordell
> > Sent: Friday, June 28, 2002 2:16 AM
> > To: Jonathan Rosenberg; sip@ietf.org
> > Subject: Re: [Sip] Proposed change in draft-ietf-sip-nat
> >
> >
> > Sounds reasonable except that to be in line with the pre-midcom / midcom
> > work all references to TURN should be to SPAN (Simple Protocol for
> > Augmenting NATs), at least for the time being.
> >
> > For those that are interested, a candidate for SPAN is at:
> >
> > http://www.ietf.org/internet-drafts/draft-cordell-midcom-span-a-00.txt
> >
> > But be advised that this is iteration 1 and so subject to change.
> >
> > Pete.
> >
> > ----- Original Message -----
> > From: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
> > To: <sip@ietf.org>
> > Sent: 28 June 2002 06:06
> > Subject: [Sip] Proposed change in draft-ietf-sip-nat
> >
> >
> > > Folks,
> > >
> > > I would like to propose a change in draft-ietf-sip-nat.
> > >
> > > Right now, this document specifies two very independent and orthogonal
> > > things. One of them is the rport parameter. This is a Via param that
> > > makes sure that the SIP response gets routed back to the source IP and
> > > port where the request came from. Without it, SIP over UDP doesn't
work
> > > through any NATs. It allows SIP to work like most other protocols in
> > > terms of its port usage. I think its uncontroversial and should move
> > > forward as is.
> > >
> > > THe other aspect is the Translate header. Its aimed at making
> > > registrations work through NAT. However, there are several problems
with
> > > the Translate header right now:
> > >
> > > 1. It "competes" with other solutions, such as STUN and TURN/SPAN,
which
> > > would allow for incoming INVITEs through NAT without any SIP specific
> > > extensions.
> > >
> > > 2. It tries to optimize for the full-cone case, by including a
parameter
> > > that indicates whether the client is behind a full-cone or symmetric
> > > NAT. I would rather not pollute SIP with this kind of NAT-specific
> > > stuff; rather, relegate it to STUN/TURN.
> > >
> > > 3. UDP registrations through NAT are really troublesome. They require
> > > very frequent re-registrations to maintain the binding (under a
minute).
> > > This poses a serious load. If you have to use UDP, a STUN or TURN
> > > solution is much better, since refreshes of the NAT bindings would
occur
> > > through those protocols, rather than SIP, and they scale much better.
> > > So, if you consider TCP (which works even better still, since you
don't
> > > need to refresh the registration in order to keep the binding active),
> > > the problem is not really translating the contact header as it is TCP
> > > connection reuse for incoming requests. Indeed, the same problem
occurs
> > > in record-routing - you want to be able to specify to your peer to
reuse
> > > the connection for incoming requests. Thats not even a nat-specific
> > > thing; its just a capability that, if we had a better story on, would
be
> > > useful for NAT.
> > >
> > >
> > > As a result, I would propose that draft-ietf-sip-nat move forward with
> > > just the rport stuff, and we look at the registration problem
> > > separately. I would like a solution for SIP that didn't depend on
> > > STUN/TURN, but I have convinced myself that this solution needs to be
> > > TCP-based, and that it is best viewed as a problem of connection
reuse.
> > > ROhan has posted a requirement draft on that topic.
> > >
> > > Flame away....
> > >
> > > -Jonathan R.
> > >
> > >
> > >
> > >
> > > --
> > > Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
> > > Chief Scientist                         First Floor
> > > dynamicsoft                             East Hanover, NJ 07936
> > > jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
> > > http://www.jdrosen.net                  PH:  (973) 952-5000
> > > http://www.dynamicsoft.com
> > >
> > >
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the application of sip
> >
> >
> >
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >



_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From daemon@optimus.ietf.org  Mon Jul  1 17:42:54 2002
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From: Tim.Moran@nokia.com
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Thread-Topic: Event Filtering
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To: <hgs@cs.columbia.edu>, <jdrosen@dynamicsoft.com>,
        <sriramp@nortelnetworks.com>, <adam@dynamicsoft.com>,
        <rsparks@dynamicsoft.com>
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	Adam/Robert/Siriram/Jonathan/Henning et. al,
	I don't watch the SIP list as much as I do the simple and sipping lists when a colleague told me of this discussion (extracted most of it from the SIP mail archive) to which I respond. I hope I haven't violated any rules by cc'ing the sipping list. 
	Before diving into the technical, would this thread be more appropriate on the SIPPING list?
	Is it possible to define a useful general-purpose filtering mechanism?  Yes - and No. If you mean one set of software that fits all - no. However, there is a good bit that can which includes: the structure, the verbs (operators) and maybe base type operations. There are some operations that don't rely on types defined in the applciation package. Time based operations are universally applicable (never say never and never say universally).
	There seems to be some interest to investigate the requirements and start the design(I'll use the term arhitecture). Architecture was chosen in the draft since we did not want to go so far as picking the implementation language (XML, SOAP, CPL, XSLT, ...). Personally, I do not care as long as it is reasonable. Admittedly, the XML we submitted in the appendix seemed like a reasonable compromise. 
	Setting a realistic scope is an admirable goal. I get a little concerned when we say minimum functionality or realistic scope given what Jonathan proposed. What is minimum for one package will not be the same for another package - necessarily. I also point out that the architecture includes mechanisms to indicate what is not supported. Nowhere is it required to implement all the functions/operators etc. Each application/package will support what it wants to. There are hooks we defined for extensibility. However, this should be the exception and not the rule.  Drawing the line for the base set of operations and extensions (as done in the draft) will be a bit tricky - but doable given the needs of the industry. For example, maybe all you need is an equality and a not operator. However, this will not satisfy the need of a watcher wanting to know when the list of buddies is greater than x or location has changed more than y. The math operators I concede are icing - but someone ma!
y prove me wrong on this. Point is, what does it hurt if we now go ahead and definea fairly complete set of verbs?
	So far, the requirements I have seen proposed are:
	Rate limiting (time based & state/attribute change based){when} and content filtering {what}.
	I suggest a BAR BOF be held in Japan to discuss this further, e.g. a design team.  Also, last I heard the moran-arch-draft was not on the agenda for Japan. Should it be?
	Regards,
	Tim Moran. Nokia Inc.

	To: Jonathan Rosenberg <jdrosen@dynamicsoft.com <mailto:jdrosen@dynamicsoft.com>> 
*	Subject: Re: [Sip] RE: Event Filtering 
*	From: Henning Schulzrinne <hgs@cs.columbia.edu <mailto:hgs@cs.columbia.edu>> 
*	Date: Sun, 30 Jun 2002 16:26:50 -0400 
*	CC: Sriram Parameswar <sriramp@nortelnetworks.com <mailto:sriramp@nortelnetworks.com>>, sip@ietf.org <mailto:sip@ietf.org> 
*	List-Id: Session Initiation Protocol <sip.ietf.org> 
*	Organization: Columbia University 
*	References: <EF1056F8EB4ED511B8FB0002A56079D401E5B0C8@zrc2c014.us.nortel.com <msg05808.html>> <3D1AC5E2.4030302@dynamicsoft.com <msg05830.html>> 
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Jonathan Rosenberg wrote:
> I acutally like the idea of a more generic filter/trigger mechanism that 
> would apply cross-packages. The tradeoff is one of generality vs. 
> complexity. The single mechanism will be more general, but probably more 
> complex than an event-specific one. How much more complex? Impossible to 
> say, of course. I generally favor generality, so if its not to omuch 
> more complex, I think its a good thing. However, unless you plan on 
> enumerating the set of filters and triggers allowed in a package, I 
> suspect that the general one won't be that much more complex. Saying 
> "only send me notifies once an hour on alternate thursdays, if the value 
> of the foo parameter is bar" is going to be complex in any case.

CPL with the calsch extensions makes expressing alternate Thursdays very 
easy :-)


> 
> I think the Moran draft is a good start, but not quite there yet. Part 
> of the problem is scope. What exactly do we wish to support, in terms of 
> filter/trigger policies? You don't want to boil the ocean with the first 
> version. I think that, if we go for a general solution, we should do 
> JUST enough to meet the *real* problems people are trying to solve, and 
> we can make it extensible to handle more down the road.
> 
> I would postulate that, realistically, the scope of triggers is going to 
> be limited to:
> 
>   * waiting for a certain event to occur within a certain state,
>   * waiting for a certain variable to take a certain value
> 
> Things like the mathematical and temporal conditions are interesting, 
> but I doubt would really be used in practice. Similarly, the filtering 
> conditions are probably limited to asking for notifications to contain 
> only certain variables.
> 
> With a well scoped set of requirements, as I have proposed above, I 
> think you can get a pretty straightforward language that would apply 
> cross-packages.
> 
> I'll also note that, if we assume that all notifications contain XML as 
> bodies (I know its not true, bear with me), then the filter could 
> possibly be an XSLT document. This might be overkill. For things that 
> are not XML, you'd need to specify an XML mapping for it to work (yuck). 
>  Just a thought.
> 
> So, I would propose that, in order to make progress on this issue, we 
> agree upon a realistic scope for these things, in terms of requirements 
> on filters and triggers, and then we can see how complex the 
> general-purpose filtering language is.

We've also looked at extending CPL for presence (draft-wu-cpl-presence, 
I thin). The question is whether the same model makes sense for events.

I don't think XSLT is the right choice. If anything, you want something 
like the XML Query language where you can formulate conditions on XML 
bodies.

Henning




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Subject: Re: [Sip] Proposed change in draft-ietf-sip-nat
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----- Original Message -----
From: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
To: "Cullen Jennings" <fluffy@cisco.com>


>
>
> Cullen Jennings wrote:
> > My read of the SPAN document was it did not even represent consensus of
> > the
> > design team much less the midcom WG.
>
> Correct.

Indeed and that's what the document says, so why the need to comment here?

To go further, I am waiting VERY patiently for comments to come in for
discussion.  Much effort has been put in on my part to reach consensus using
many small steps to try and bring the team along as a whole and I will
continue to work with that goal in mind.  Rest assured that this does not
mean imposing my will on others, but where I believe I have valid technical
issues, I will present them and look forward to them being discussed in an
open (and technical) way.

> >
> > Why should references to TURN or SPAM be changed to just SPAN?
>
> More importantly, this issue is totally irrelevant to this thread.
> Draft-ietf-sip-nat will have references to neither TURN, nor SPAN, nor
> STUN. Connection reuse has nothing to do with it either. THe only place
> TURN/SPAN/STUN should show up is the nat-scenarios document. That one is
> on the backburner until these issues are sorted out, and is not what
> this thread is about in any case.

More than likely.  So why not acknowledge the editorial error (which
requires no action) and move on.

>
> -Jonathan R.
>
>
> --
> Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
> Chief Scientist                         First Floor
> dynamicsoft                             East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
> http://www.jdrosen.net                  PH:  (973) 952-5000
> http://www.dynamicsoft.com



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From daemon@optimus.ietf.org  Mon Jul  1 18:46:15 2002
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I tend to agree with Henning that if we focus
on a solution for XML documents, then XML Query
seems like the best candidate. The million
dollar question is whether this is an appropriate
scope of the problem. A BAR BOF on the topic
seems like a good idea. Please include me on 
any scheduling e-mails for this :)

Cheers
sean

> 	I suggest a BAR BOF be held in Japan to discuss
> this further, e.g. a design team.  Also, last I
> heard the moran-arch-draft was not on the agenda for
> Japan. Should it be?
> 	Regards,
> 	Tim Moran. Nokia Inc.

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From daemon@optimus.ietf.org  Tue Jul  2 00:47:48 2002
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From: "Ranjit Avasarala" <ranjit.avasarala@bplmail.com>
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Subject: [Sip] query on NOTIFY syntax
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Hi
     as per draft-ietf-sip-refer-05.txt , the Content Type in NOTIFY is
message/sipfrag and contains the header Subscription-State

But as per draft-rosenberg-sip-reg-00)1).txt, the Content Type in NOTIFY is
application/reginfo+xml

which one should be used?

Actually my requirement is while implementing the feature of automatic
callback by user A. user A subscribes to this feature and user A should be
notified by the server when user B is free.

I use Event: autoCallBackUserB

In NOTIFY: subscription-state: "init" / "active"

TIA
Ranjit




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From daemon@optimus.ietf.org  Tue Jul  2 01:07:44 2002
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Chris Martin wrote:
> Jonathan Rosenberg wrote:

> To clarify the latter comment SIP traversal solutions for the enterprise
> that involves keep alives or topology discovery on enterprise firewalls
> may be considered a violation of enterprise security policy, not to
> mention the excess traffic, which is typically hunted down to be
> silenced on many of these types of networks, and lastly the potential
> risk of providing additional vulnerabilities to the internal network
> that are yet unknown. Many infrastructures do not want internal users
> having knowlege of the topology. Placing the ability to do so on a
> client to determine this information may be frowned upon as well.

I can only assume you are talking about STUN. STUN doesn't discover 
topology, it just determines whether or not there exist one or more NATs 
between the client and the public Internet.


> 
> Anyway the firewall vendors have begun to provide SIP solutions to
> address this now, so it is becoming a moot point. MIDCOM may provide a
> way to offer additional support for many items other than firewalls,
> such as load balancing devices etc, so this has much merit, but a
> traversal solution, IMHO, is not necessary or acceptable, except in the
> SOHO realm, if that. 

I disagree.

You are assuming that the only solution an enterprise would find viable 
is an ALG or a midcom-controllable firewall. I think it is legitimate 
for an enterprise to decide to deploy SIP using one of several other 
approaches, all of which are documented in 
draft-ietf-sipping-nat-scenarios. One of these, for example, involves 
the deployment of a SIP server in the DMZ, along with a configuration 
change in the firewall. I claim that it should be possible to configure 
the firewall to support SIP without requiring that configuration to be 
excessively permissive; this basically means that SIP should be workable 
with client initiated "connections", much like many other application 
protocols, and that is the aim of this draft.

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com


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Hi Bert,

Great mail! Thanks!

> >However, there is no specific "remote ringback session" in
> >SIP. You establish a generic early dialiog, by sending an
> >offer and receiving an answer,
> >
> >[DA]As the offer/answer draft says: one "MUST be prepared to
> >receive media" after the sending of the offer, so it does not
> >require for UAC to receive an answer in order to start the
> >media playback (as the early media, in my understanding,
> >always come from the UAS i.e. ringback tone, announcement,...
> >and not other way around)
>
>
> We want the UAC to render received media once it's sent an offer to
> prevent clipping.  We need this, it causes an unacceptable user
> experience if we don't do this.  In addition, we don't want to require
> the UAS to provide "call progress" media, we have the 180 for it to
> indicate the callee is being alerted, the UAC can convey this state to
> the caller through local ringback or whatever.  It's been
> proposed/discussed/and I believe there's consensus that a UAS will send
> 183 (or 180 recently) with SDP when it intends to present media that
> should be rendered to the caller.  (Now this is not required of the UAS
> but it is "polite" in that it informs the UAC of its intentions.)  The
> expected UAC behavior is for it to stop "ringback" indication at this
> point.

[CHH] True. But, this is the case where the UAC first generates local
ringback, then starts receiving remote early media, and the case
I've brought up is the oposit.

Also, what you wrote was very much true until the media mode indicators
(sendrecv, inactive etc) were introduced to SIP. Now, if a UAC sends an
INVITE with "inactive", is it still a MUST for the UAC to be able to receive
media at that point? One of the reason to use "inactive" was to PREVENT
early media, so...

> Now there's the case when the UAS is doing something more with the
> session, and stops sending media to the UAC.  In this case, the caller
> will not see/hear any "call progress".  What's being asked for now is
> clarification on how to "ask" the UAC to again generate call progress
> indication.  It's been suggested that another 18x without SDP indicate
> this scenario.  As Christer indicated, there are some I-Ds that describe
> this behavior.

[CHH] To clarify, the drafts I refer to is bis-09 and the ISUP/SIP mapping
draft.

> There was a period in time where there seemed to be consensus in the WG on
> this.  However, offer-answer, unify, ... I-Ds
> have since come along that more rigorously define "early dialogs",
> meaning of SDP, etc.  And again, the presence of SDP in 18x has nothing
> to do with when a UAC should generate or expect session progress
> indications.

[CHH] True, and that is the reason I asked if the SDP as an ringback
indicator (which it IS used as in some drafts) is a good idea in the first
place. Like Bert said, those drafts were written long before anyone had
heard of any offer/answer rules etc...

> AFAIK, it's not possible in SIP to indicate to a UAC that
> it should stop rendering media and go back to local generation of
> progress indication.  Nor do we want to try to do this in SIP.

[CHH] If that is the case, the text in the drafts I refered to should be
rewritten. I guess what you propose is:

"When the UAC receives the 180 it does the ringback decission based on if
there is an early media session established, and the state of it (sendrecv,
inactive,...) - NOT based on if there is a SPD in the 180 or not (of course,
the SDP in 180 may be the SDP used to establish the early media session, but
that is a separate issue)."

> IMHO, services/applications that will generate early media and then stop
> have the responsibility to make the caller's user experience acceptable
> (i.e., by playing an announcement, music, or whatever).  In Christer's
> B2BUA example, the B2BUA's UAC assumes the responsibility to provide
> call progress indication to its user (which is another UAC).

[CHH] If that's fine with those who are going to build/use such
applications, it's fine with me :) I assumed, however, that people don't
want the B2BUA to generate ringback tones on behalf of the SIP phones...
And, I also think it could be a relatively common SIP application, so I
don't think we're talking about a strange and weird scenarios...

> We should only recommend that a UAC generate a local indication upon
> receipt of a 180 and then stop upon receipt of media (which it should
> render).  Services such as Christer's example must take it upon
> themselves to provide an acceptable user experience.  Otherwise we add
> complexity to the protocol and degrade both interoperability and user
> experience.

[CHH] I don't think the UAC shall "stop upon receipt of media". If the UAS
starts sending media, it shall also send a 18x with an answer to the offer
sent in INVITE to establish an early media session (or, an UPDATE it the
early session already exists, but in "inactive" mode).

I also would like to get a clarification if the offer/answer rules in bis-09
"allow" us to establish an early media session by sending the answer in an
unreliable 18x?

> This is what I'd propose be the "official" clarification.  My 2 cents
> anyway.

[CHH] That's what I asked for :)

Regards,

Christer Holmberg
Ericsson Finland




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--NextPart

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

	Title		: The Stream Control Transmission Protocol as a 
                          Transport for  for the Session Initiation Protocol
	Author(s)	: J. Rosenberg, H. Schulzrinne, G. Camarillo
	Filename	: draft-ietf-sip-sctp-03.txt
	Pages		: 8
	Date		: 01-Jul-02
	
This document specifies a mechanism for usage of SCTP (the Stream
Control Transmission Protocol) as the transport between SIP entities.
SCTP is a new protocol which provides several features that may prove
beneficial for transport between SIP entities which exchange a large
amount of messages, including gateways and proxies. As SIP is
transport independent, support of SCTP is a relatively
straightforward process, nearly identical to support for TCP.

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From daemon@optimus.ietf.org  Tue Jul  2 08:37:30 2002
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From: Saroj Kumar <sksaroj@yahoo.com>
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Subject: [Sip] Question on SIP
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I have the following data.

User name: 	5088855003
Password: 	5088855003
Realm :  	BroadWorks
Uri: 	 	sip:10.15.66.133
Algorithm:	MD5
Nonce:		1024027759288

I am using the following code for calculation of
Response.
  char * pszNonce = "1024027759288";
  char * pszCNonce = "";	
  char * pszUser = "5088855003";
  char * pszRealm = "BroadWorks";
  char * pszPass = "5088855003";
  char * pszAlg = "md5";
  char szNonceCount[9] = "";
  char * pszMethod ="Get;
  char * pszQop ="";
  char * pszURI ="sip:10.15.66.133";
  HASHHEX HA1;
  HASHHEX HA2 = "";
  HASHHEX Response;

 DigestCalcHA1(pszAlg, pszUser, pszRealm, pszPass,
pszNonce,pszCNonce, HA1);
 DigestCalcResponse(HA1, pszNonce, szNonceCount,
pszCNonce, pszQop,
 pszMethod, pszURI, HA2, Response);
  printf("Response = %s\n", Response);

and getting the out put as
725a9146fa0eb8bd8fd709086f190c01
but I am expecting the result as
b0f6dbce82b1fe5f9adf5d124179ec2a.

Whatever examples I got in the net cNonce value is
used, but in this case I don't have the cNonce as
there is no qop.


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From daemon@optimus.ietf.org  Tue Jul  2 14:59:24 2002
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This is a very interesting problem. Should the requirements
include the ability to deal with intermediaries such as
NAT/firewalls that may not strictly be SIP entities? Also, I 
wonder if the proxy-proxy connection problem is as critical
as the UAC-proxy connection reuse problem. Might SCTP be a
better solution for proxy-proxy connection management?

Thanks,

Sean Olson
Microsoft
http://www.microsoft.com/rtc



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From daemon@optimus.ietf.org  Tue Jul  2 15:14:25 2002
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Subject: RE: [Sip] Local vs Remote ringback on 180
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Hi Christer,

more of my comments below...

Christer Holmberg wrote:
>
> Hi Bert,
>
> Great mail! Thanks!
>
> > >However, there is no specific "remote ringback session" in
> > >SIP. You establish a generic early dialiog, by sending an
> > >offer and receiving an answer,
> > >
> > >[DA]As the offer/answer draft says: one "MUST be prepared to
> > >receive media" after the sending of the offer, so it does not
> > >require for UAC to receive an answer in order to start the
> > >media playback (as the early media, in my understanding,
> > >always come from the UAS i.e. ringback tone, announcement,...
> > >and not other way around)
> >
> >
> > We want the UAC to render received media once it's sent an offer to
> > prevent clipping.  We need this, it causes an unacceptable user
> > experience if we don't do this.  In addition, we don't want
> to require
> > the UAS to provide "call progress" media, we have the 180 for it to
> > indicate the callee is being alerted, the UAC can convey
> this state to
> > the caller through local ringback or whatever.  It's been
> > proposed/discussed/and I believe there's consensus that a
> UAS will send
> > 183 (or 180 recently) with SDP when it intends to present media that
> > should be rendered to the caller.  (Now this is not
> required of the UAS
> > but it is "polite" in that it informs the UAC of its
> intentions.)  The
> > expected UAC behavior is for it to stop "ringback"
> indication at this
> > point.
>
> [CHH] True. But, this is the case where the UAC first generates local
> ringback, then starts receiving remote early media, and the case
> I've brought up is the oposit.
>
> Also, what you wrote was very much true until the media mode
> indicators
> (sendrecv, inactive etc) were introduced to SIP. Now, if a
> UAC sends an
> INVITE with "inactive", is it still a MUST for the UAC to be
> able to receive
> media at that point? One of the reason to use "inactive" was
> to PREVENT
> early media, so...

Yes, the ability to now use "inactive" changes things.  But if a UAC
indicates "inactive" then there is no means to send the corresponding
media streams.  And the caller then can not see/hear far-end media-based
progress indications.

>
> > Now there's the case when the UAS is doing something more with the
> > session, and stops sending media to the UAC.  In this case,
> the caller
> > will not see/hear any "call progress".  What's being asked
> for now is
> > clarification on how to "ask" the UAC to again generate
> call progress
> > indication.  It's been suggested that another 18x without
> SDP indicate
> > this scenario.  As Christer indicated, there are some I-Ds
> that describe
> > this behavior.
>
> [CHH] To clarify, the drafts I refer to is bis-09 and the
> ISUP/SIP mapping
> draft.
>
> > There was a period in time where there seemed to be
> consensus in the WG on
> > this.  However, offer-answer, unify, ... I-Ds
> > have since come along that more rigorously define "early dialogs",
> > meaning of SDP, etc.  And again, the presence of SDP in 18x
> has nothing
> > to do with when a UAC should generate or expect session progress
> > indications.
>
> [CHH] True, and that is the reason I asked if the SDP as an ringback
> indicator (which it IS used as in some drafts) is a good idea
> in the first
> place. Like Bert said, those drafts were written long before
> anyone had
> heard of any offer/answer rules etc...

I've glanced through bis-09 for a discussion on UAC behavior for 18x and
didn't find anything on when or when not to generate ringback locally.
The SIP/ISUP I-D describes this for PSTN gateways.  Being that that I-D
is dealing with PSTN interworking (where the far end typically provides
audible call progress indications) a convention is needed that addresses
when local vs. far-end call progress indication is needed.  I'm a little
behind in my SIP-ISUP reading but I think the precedures described
therein apply to PSTN gateways and not general SIP UAs.  Also, PSTN
interworking is a distinct enough scenario that additional semantics
(for 180 vs. 183 and presence of SDP) are OK (but only as far as
gateways are concerned).

>
> > AFAIK, it's not possible in SIP to indicate to a UAC that
> > it should stop rendering media and go back to local generation of
> > progress indication.  Nor do we want to try to do this in SIP.
>
> [CHH] If that is the case, the text in the drafts I refered
> to should be
> rewritten. I guess what you propose is:
>
> "When the UAC receives the 180 it does the ringback decission
> based on if
> there is an early media session established, and the state of
> it (sendrecv,
> inactive,...) - NOT based on if there is a SPD in the 180 or
> not (of course,
> the SDP in 180 may be the SDP used to establish the early
> media session, but
> that is a separate issue)."

Again, I didn't find this kind of discussion in bis-09.  I myself think
some guidance (as you suggest above) would be helpful in a some I-D.
Otherwise we have to debate this aspect on mailing lists :)

>
> > IMHO, services/applications that will generate early media
> and then stop
> > have the responsibility to make the caller's user
> experience acceptable
> > (i.e., by playing an announcement, music, or whatever).  In
> Christer's
> > B2BUA example, the B2BUA's UAC assumes the responsibility to provide
> > call progress indication to its user (which is another UAC).
>
> [CHH] If that's fine with those who are going to build/use such
> applications, it's fine with me :) I assumed, however, that
> people don't
> want the B2BUA to generate ringback tones on behalf of the
> SIP phones...
> And, I also think it could be a relatively common SIP
> application, so I
> don't think we're talking about a strange and weird scenarios...

Well, I think the B2BUA has to "manage" the callee in this case, given
the absence within the protocol to tell a UAC to generate ringback (in
the general case).  I think it would be a useful capability but doubt
attaching additional semantics to SDP presence and/or 180/183 is the way
to do it.  Currently it's only a MAY in bis-09 that a UAC generates
local ringback upon receipt of a 180 anyway.

>
> > We should only recommend that a UAC generate a local indication upon
> > receipt of a 180 and then stop upon receipt of media (which
> it should
> > render).  Services such as Christer's example must take it upon
> > themselves to provide an acceptable user experience.
> Otherwise we add
> > complexity to the protocol and degrade both
> interoperability and user
> > experience.
>
> [CHH] I don't think the UAC shall "stop upon receipt of
> media". If the UAS
> starts sending media, it shall also send a 18x with an answer
> to the offer
> sent in INVITE to establish an early media session (or, an
> UPDATE it the
> early session already exists, but in "inactive" mode).

Again, AFAIK a UAS can send media to the UAC after receiving the UAC's
SDP.  No response is required to be sent back to the UAC.  Sending a
response with SDP is the "polite" thing to do IMHO as it's an explicit
indication that the UAS will send media (and allows RTP/RTCP to "work").

I believe UAC behavior here is an largely an implementation issue, and
what I said are my views on what will provide an acceptable user
experience in the general case.  If the SIP WG wants to define
"operational semantics" for some protocol scenarios I'm OK with that.
The subject of this thread is a good one IMHO, and I remember a time
when this was discussed.

>
> I also would like to get a clarification if the offer/answer
> rules in bis-09
> "allow" us to establish an early media session by sending the
> answer in an
> unreliable 18x?

My read indicates answers MUST be in a reliable response, which is only
200 in bis-09.  This is stated pretty clearly IMHO.  I'm not sure what a
"early media session" is but if you want to complete the offer-answer
exchange prior to sending/receiving a 200 OK, then there's the 100-rel
extension (as I'm sure you're aware :).

Best regards,
Bert

>
> > This is what I'd propose be the "official" clarification.
> My 2 cents
> > anyway.
>
> [CHH] That's what I asked for :)
>
> Regards,
>
> Christer Holmberg
> Ericsson Finland
>
>
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>


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From daemon@optimus.ietf.org  Tue Jul  2 15:36:00 2002
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From: Adam Roach <adam@dynamicsoft.com>
To: "'Ranjit Avasarala'" <ranjit.avasarala@bplmail.com>,
        sip-ietf
	 <sip@ietf.org>
Subject: RE: [Sip] query on NOTIFY syntax
Date: Tue, 2 Jul 2002 14:52:51 -0400 
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The content type depends on the package being used. draft-ietf-sip-refer-05
defines one package; draft-rosenberg-sip-reg-01 defines a different one.

Your proposal to use a brand-new and feature-specific event package
of "autoCallBackUserB" is unlikely to get any traction at all.
There is, however, an event package that is being defined that
includes all the functionality you need for an auto-callback feature.
See draft-ietf-sipping-dialog-package-00.

/a

> -----Original Message-----
> From: Ranjit Avasarala [mailto:ranjit.avasarala@bplmail.com]
> Sent: Monday, July 01, 2002 23:25
> To: sip-ietf
> Subject: [Sip] query on NOTIFY syntax
> 
> 
> Hi
>      as per draft-ietf-sip-refer-05.txt , the Content Type in 
> NOTIFY is
> message/sipfrag and contains the header Subscription-State
> 
> But as per draft-rosenberg-sip-reg-00)1).txt, the Content 
> Type in NOTIFY is
> application/reginfo+xml
> 
> which one should be used?
> 
> Actually my requirement is while implementing the feature of automatic
> callback by user A. user A subscribes to this feature and 
> user A should be
> notified by the server when user B is free.
> 
> I use Event: autoCallBackUserB
> 
> In NOTIFY: subscription-state: "init" / "active"
> 
> TIA
> Ranjit
> 
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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From daemon@optimus.ietf.org  Wed Jul  3 03:43:28 2002
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Hi!

A small clarification question on generation of To tags:

Bis-09 says that responses can be sent stateless (ie no transaction is
created), but that retransmits of the request MUST create the same
response every time - including the To tag.

Now, one way to achieve this is to calculate the To tag as a hash from
some headers in the request.

However, chapter 19.3 in bis-09 says that the tag must be
"cryptographically random with at least 32 bits of randomness".

To me that means that we must also use some kind of random number when
we
create the hash, but how are we now going to make sure the result is the

same every time (even if the header hash will be the same the random
number will be different every time)???

Regards,

Christer Holmberg
Ericsson Finland



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From daemon@optimus.ietf.org  Wed Jul  3 05:41:36 2002
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Hi Jonathan

As per draft the MIME type in NOTIFY should be application/reginfo+xml.
but in draft on REFER MIME type for NOTIFy is given as message/sipfrag.

so which one should be used?

in my case say I want to register a particular event in the server, so can I
use SUBSCRIBE / NOTIFY pair?
with just plain mesaage and no xml.

Thanks
Ranjit





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Date: Wed, 03 Jul 2002 18:10:06 +0900
From: Eiji Tomimura <tomimura@sei.co.jp>
To: <sip@ietf.org>
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Dear all,

I have a question regarding ABNF for SIP.

In bis-09, it defines that :
    rfc1123-date = wkday "," date1 SP time SP "GMT"

Though, in RFC2616 "HTTP/1.1", it defines that:
    rfc1123-date = wkday "," SP date1 SP time SP "GMT"

Is it intentional? Or, is it a bug of either side?

Becauses our SIP layer is based on RFC2616 format, it reject Date header
follows the format of bis-09 as syntax error. So I'm wondering which format
should we use.

Thanks in advance.

Regards,
- Eiji Tomimura

-- 
Eiji Tomimura 
Senior Engineer 
Communications Technology Dept., Information Technology R&D Laboratories
Sumitomo Electric Industries, Ltd.
Phone: +81-6-6466-5643  Fax: +81-6-6466-5732  Email: tomimura@sei.co.jp


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From daemon@optimus.ietf.org  Wed Jul  3 07:23:32 2002
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Subject: [Sip] I-D ACTION:draft-mahy-sip-join-and-fork-01.txt
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: The Session Inititation Protocol (SIP) 'Join' Header
	Author(s)	: R. Mahy, D. Petrie
	Filename	: draft-mahy-sip-join-and-fork-01.txt
	Pages		: 14
	Date		: 02-Jul-02
	
This document defines a new header for use with SIP multi-party
applications and call control.  The Join header is used to logically
join an existing SIP dialog with a new SIP dialog.  This primitive
can be used to enable a variety of features, for example: 'Barge-In',
answering-machine-style 'Message Screening' and 'Call Center
Monitoring'.  Note that definition of these example features is non-
normative.

A URL for this Internet-Draft is:
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Date: Wed, 3 Jul 2002 06:15:01 -0500
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Thread-Topic: [Sip] Question about ABNF for SIP
Thread-Index: AcIic6ir/2YaSDwgTQGvom8tNEuPVAADxWKQ
From: "Rob Phillips" <rob@netrake.com>
To: <sip@ietf.org>
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The problem is with the bis-09 definition.  Check the examples in bis-09: from those, it's obvious the rule should be:

	rfc1123-date = wkday "," SP date1 SP time SP "GMT"

- rob

-----Original Message-----
From: Eiji Tomimura [mailto:tomimura@sei.co.jp]
Sent: Wednesday, July 03, 2002 4:10 AM
To: sip@ietf.org
Subject: [Sip] Question about ABNF for SIP


Dear all,

I have a question regarding ABNF for SIP.

In bis-09, it defines that :
    rfc1123-date = wkday "," date1 SP time SP "GMT"

Though, in RFC2616 "HTTP/1.1", it defines that:
    rfc1123-date = wkday "," SP date1 SP time SP "GMT"

Is it intentional? Or, is it a bug of either side?

Becauses our SIP layer is based on RFC2616 format, it reject Date header
follows the format of bis-09 as syntax error. So I'm wondering which format
should we use.

Thanks in advance.

Regards,
- Eiji Tomimura

-- 
Eiji Tomimura 
Senior Engineer 
Communications Technology Dept., Information Technology R&D Laboratories
Sumitomo Electric Industries, Ltd.
Phone: +81-6-6466-5643  Fax: +81-6-6466-5732  Email: tomimura@sei.co.jp


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@ns.ietf.org  Wed Jul  3 08:30:53 2002
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Date: Tue, 02 Jul 2002 22:03:09 -0500
From: Chris Martin <chris@infravast.com>
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CC: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
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        sip@ietf.org
Subject: Re: [Sip] refreshing NAT binding for SIP with UDP registrations
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Chris Martin wrote:
> 
> Jonathan Rosenberg wrote:
> >
> > Chris Martin wrote:
> > > Jonathan Rosenberg wrote:
> >
> > > To clarify the latter comment SIP traversal solutions for the enterprise
> > > that involves keep alives or topology discovery on enterprise firewalls
> > > may be considered a violation of enterprise security policy, not to
> > > mention the excess traffic, which is typically hunted down to be
> > > silenced on many of these types of networks, and lastly the potential
> > > risk of providing additional vulnerabilities to the internal network
> > > that are yet unknown. Many infrastructures do not want internal users
> > > having knowlege of the topology. Placing the ability to do so on a
> > > client to determine this information may be frowned upon as well.
> >
> > I can only assume you are talking about STUN. STUN doesn't discover
> > topology, it just determines whether or not there exist one or more NATs
> > between the client and the public Internet.
> 
> I am not focusing on one implementation over another...many of them
> provide this level of vulnerability. A client that uses mechanisms to
> ascertain information of topology is an additional risk that many
> enterprise, or for that matter any responsible network administrator,
> would permit to be implemented.
       ^ Would not permit
> 
> >
> > >
> > > Anyway the firewall vendors have begun to provide SIP solutions to
> > > address this now, so it is becoming a moot point. MIDCOM may provide a
> > > way to offer additional support for many items other than firewalls,
> > > such as load balancing devices etc, so this has much merit, but a
> > > traversal solution, IMHO, is not necessary or acceptable, except in the
> > > SOHO realm, if that.
> >
> > I disagree.
> >
> > You are assuming that the only solution an enterprise would find viable
> > is an ALG or a midcom-controllable firewall. I think it is legitimate
> > for an enterprise to decide to deploy SIP using one of several other
> > approaches, all of which are documented in
> > draft-ietf-sipping-nat-scenarios. One of these, for example, involves
> > the deployment of a SIP server in the DMZ, along with a configuration
> > change in the firewall.
> 
> Why deploy a SIP proxy in the DMZ if you can deploy or upgrade a
> firewall that provides SIP functionality? The solutions being devloped
> today range from ALG's to B2BUA type firewall solutions. In many cases
> the B2BUA's can also act as standalone proxy/registrars at a minimum
> cost, and they are capable of interoperability with external SIP
> proxies.
> 
> > I claim that it should be possible to configure
> > the firewall to support SIP without requiring that configuration to be
> > excessively permissive
> 
> Thats exactly what I said about what the firewall vendors are doing...it
> is not complicated and it is not overly permissive, and it does not
> require additional physical proxy in the DMZ to provide the desired
> functionality.
> 
> Hence, the SIP enabled firewalls that are being developed and deployed
> currently by well known and emerging vendors. These solutions provide
> far better solutions that fit the enterprise model than those that
> provide a discovery or traversal mechanism or the requirement of
> additional boxes on the enterprise..
> 
> >; this basically means that SIP should be workable
> > with client initiated "connections", much like many other application
> > protocols, and that is the aim of this draft.
> 
> Nothing like adding a denial of service tool to all of ones internal
> clients.
> 
> >
> > -Jonathan R.
> > --
> > Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
> > Chief Scientist                         First Floor
> > dynamicsoft                             East Hanover, NJ 07936
> > jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
> > http://www.jdrosen.net                  PH:  (973) 952-5000
> > http://www.dynamicsoft.com


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@ns.ietf.org  Wed Jul  3 08:35:39 2002
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Eiji Tomimura'" <tomimura@sei.co.jp>, sip@ietf.org
Subject: RE: [Sip] Question about ABNF for SIP
Date: Wed, 3 Jul 2002 08:00:16 -0400 
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This problem has been reported before.  We fixed it in the
text of the forthcoming RFC.  The SP is added.

Brian

> -----Original Message-----
> From: Eiji Tomimura [mailto:tomimura@sei.co.jp]
> Sent: Wednesday, July 03, 2002 5:10 AM
> To: sip@ietf.org
> Subject: [Sip] Question about ABNF for SIP
> 
> 
> Dear all,
> 
> I have a question regarding ABNF for SIP.
> 
> In bis-09, it defines that :
>     rfc1123-date = wkday "," date1 SP time SP "GMT"
> 
> Though, in RFC2616 "HTTP/1.1", it defines that:
>     rfc1123-date = wkday "," SP date1 SP time SP "GMT"
> 
> Is it intentional? Or, is it a bug of either side?
> 
> Becauses our SIP layer is based on RFC2616 format, it reject 
> Date header
> follows the format of bis-09 as syntax error. So I'm 
> wondering which format
> should we use.
> 
> Thanks in advance.
> 
> Regards,
> - Eiji Tomimura
> 
> -- 
> Eiji Tomimura 
> Senior Engineer 
> Communications Technology Dept., Information Technology R&D 
> Laboratories
> Sumitomo Electric Industries, Ltd.
> Phone: +81-6-6466-5643  Fax: +81-6-6466-5732  Email: 
> tomimura@sei.co.jp
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@ns.ietf.org  Wed Jul  3 09:24:42 2002
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Date: Tue, 02 Jul 2002 21:41:56 -0500
From: Chris Martin <chris@infravast.com>
Reply-To: matrix@dasecurenetworks.com
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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
CC: matrix@dasecurenetworks.com, Sean Olson <seancolson@yahoo.com>,
        Cullen Jennings <fluffy@cisco.com>, sip@ietf.org
Subject: Re: [Sip] refreshing NAT binding for SIP with UDP registrations
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Jonathan Rosenberg wrote:
> 
> Chris Martin wrote:
> > Jonathan Rosenberg wrote:
> 
> > To clarify the latter comment SIP traversal solutions for the enterprise
> > that involves keep alives or topology discovery on enterprise firewalls
> > may be considered a violation of enterprise security policy, not to
> > mention the excess traffic, which is typically hunted down to be
> > silenced on many of these types of networks, and lastly the potential
> > risk of providing additional vulnerabilities to the internal network
> > that are yet unknown. Many infrastructures do not want internal users
> > having knowlege of the topology. Placing the ability to do so on a
> > client to determine this information may be frowned upon as well.
> 
> I can only assume you are talking about STUN. STUN doesn't discover
> topology, it just determines whether or not there exist one or more NATs
> between the client and the public Internet.

I am not focusing on one implementation over another...many of them
provide this level of vulnerability. A client that uses mechanisms to
ascertain information of topology is an additional risk that many
enterprise, or for that matter any responsible network administrator,
would permit to be implemented.

> 
> >
> > Anyway the firewall vendors have begun to provide SIP solutions to
> > address this now, so it is becoming a moot point. MIDCOM may provide a
> > way to offer additional support for many items other than firewalls,
> > such as load balancing devices etc, so this has much merit, but a
> > traversal solution, IMHO, is not necessary or acceptable, except in the
> > SOHO realm, if that.
> 
> I disagree.
> 
> You are assuming that the only solution an enterprise would find viable
> is an ALG or a midcom-controllable firewall. I think it is legitimate
> for an enterprise to decide to deploy SIP using one of several other
> approaches, all of which are documented in
> draft-ietf-sipping-nat-scenarios. One of these, for example, involves
> the deployment of a SIP server in the DMZ, along with a configuration
> change in the firewall. 

Why deploy a SIP proxy in the DMZ if you can deploy or upgrade a
firewall that provides SIP functionality? The solutions being devloped
today range from ALG's to B2BUA type firewall solutions. In many cases
the B2BUA's can also act as standalone proxy/registrars at a minimum
cost, and they are capable of interoperability with external SIP
proxies.

> I claim that it should be possible to configure
> the firewall to support SIP without requiring that configuration to be
> excessively permissive

Thats exactly what I said about what the firewall vendors are doing...it
is not complicated and it is not overly permissive, and it does not
require additional physical proxy in the DMZ to provide the desired
functionality.

Hence, the SIP enabled firewalls that are being developed and deployed
currently by well known and emerging vendors. These solutions provide
far better solutions that fit the enterprise model than those that
provide a discovery or traversal mechanism or the requirement of
additional boxes on the enterprise..

>; this basically means that SIP should be workable
> with client initiated "connections", much like many other application
> protocols, and that is the aim of this draft.

Nothing like adding a denial of service tool to all of ones internal
clients.

> 
> -Jonathan R.
> --
> Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
> Chief Scientist                         First Floor
> dynamicsoft                             East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
> http://www.jdrosen.net                  PH:  (973) 952-5000
> http://www.dynamicsoft.com


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@ns.ietf.org  Wed Jul  3 09:41:01 2002
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Date: Wed, 03 Jul 2002 16:09:58 +0300
From: Christer Holmberg <christer.holmberg@lmf.ericsson.se>
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To: Bert Culpepper <bert.culpepper@intervoice-brite.com>
CC: "'Alexeitsev, D'" <D.Alexeitsev@telekom.de>, sip@ietf.org
Subject: Re: [Sip] Local vs Remote ringback on 180
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Hi Bert,

As a conclusion to this discussion my proposal for a clarification would be
that the local ringback generation depends on if a early media session has
been established or not when the 180 arrives.

And, IF a UAS sends early media, and then wants the UAC to generate local
ringback, it first have to put the early media on hold, using UPDATE, then
send 180 (the UAC MAY now generate local ringback, since there is no early
media), and when the called party answers the UAS will first send a new
UPDATE, to un-hold the media, then send 200 for the INVITE. I still think
this process is very messy, and "much pain for little gain", but since
that's the only way SIP allows us to do it I am not going to argue against
it :)

I still would like to get a clarification on the
"UPDATE-and-buffering-of-200) issue I brought up, though :) Personally I see
no reason why to buffer the 200, but the UAS may have to buffer the second
UPDATE (used to un-hold the media) until it has completed the first UPDATE
(used to put media on hold), which means there is again a chance of media
clipping.

Now, as Bert mentioned, the UAS may start sending early media without
indicating it (sending 183 with SDP), and then still send 180 without SDP
even if it sends early media, so in that case I guess there is a risk the
local ringback WILL get mixed with the early media (but, we can get early
media mixing in forking scenarios, anyway) - UNLESS the UAC is able to do
some decissions based on detected media (I don't think that would work very
well, but it's up to each implementor to decide for him/herself).

Or, does anyone have another proposal?

Thanks!

Regards,

Christer Holmberg
Ericsson Finland






Bert Culpepper wrote:

> Hi Christer,
>
> more of my comments below...
>
> Christer Holmberg wrote:
> >
> > Hi Bert,
> >
> > Great mail! Thanks!
> >
> > > >However, there is no specific "remote ringback session" in
> > > >SIP. You establish a generic early dialiog, by sending an
> > > >offer and receiving an answer,
> > > >
> > > >[DA]As the offer/answer draft says: one "MUST be prepared to
> > > >receive media" after the sending of the offer, so it does not
> > > >require for UAC to receive an answer in order to start the
> > > >media playback (as the early media, in my understanding,
> > > >always come from the UAS i.e. ringback tone, announcement,...
> > > >and not other way around)
> > >
> > >
> > > We want the UAC to render received media once it's sent an offer to
> > > prevent clipping.  We need this, it causes an unacceptable user
> > > experience if we don't do this.  In addition, we don't want
> > to require
> > > the UAS to provide "call progress" media, we have the 180 for it to
> > > indicate the callee is being alerted, the UAC can convey
> > this state to
> > > the caller through local ringback or whatever.  It's been
> > > proposed/discussed/and I believe there's consensus that a
> > UAS will send
> > > 183 (or 180 recently) with SDP when it intends to present media that
> > > should be rendered to the caller.  (Now this is not
> > required of the UAS
> > > but it is "polite" in that it informs the UAC of its
> > intentions.)  The
> > > expected UAC behavior is for it to stop "ringback"
> > indication at this
> > > point.
> >
> > [CHH] True. But, this is the case where the UAC first generates local
> > ringback, then starts receiving remote early media, and the case
> > I've brought up is the oposit.
> >
> > Also, what you wrote was very much true until the media mode
> > indicators
> > (sendrecv, inactive etc) were introduced to SIP. Now, if a
> > UAC sends an
> > INVITE with "inactive", is it still a MUST for the UAC to be
> > able to receive
> > media at that point? One of the reason to use "inactive" was
> > to PREVENT
> > early media, so...
>
> Yes, the ability to now use "inactive" changes things.  But if a UAC
> indicates "inactive" then there is no means to send the corresponding
> media streams.  And the caller then can not see/hear far-end media-based
> progress indications.
>
> >
> > > Now there's the case when the UAS is doing something more with the
> > > session, and stops sending media to the UAC.  In this case,
> > the caller
> > > will not see/hear any "call progress".  What's being asked
> > for now is
> > > clarification on how to "ask" the UAC to again generate
> > call progress
> > > indication.  It's been suggested that another 18x without
> > SDP indicate
> > > this scenario.  As Christer indicated, there are some I-Ds
> > that describe
> > > this behavior.
> >
> > [CHH] To clarify, the drafts I refer to is bis-09 and the
> > ISUP/SIP mapping
> > draft.
> >
> > > There was a period in time where there seemed to be
> > consensus in the WG on
> > > this.  However, offer-answer, unify, ... I-Ds
> > > have since come along that more rigorously define "early dialogs",
> > > meaning of SDP, etc.  And again, the presence of SDP in 18x
> > has nothing
> > > to do with when a UAC should generate or expect session progress
> > > indications.
> >
> > [CHH] True, and that is the reason I asked if the SDP as an ringback
> > indicator (which it IS used as in some drafts) is a good idea
> > in the first
> > place. Like Bert said, those drafts were written long before
> > anyone had
> > heard of any offer/answer rules etc...
>
> I've glanced through bis-09 for a discussion on UAC behavior for 18x and
> didn't find anything on when or when not to generate ringback locally.
> The SIP/ISUP I-D describes this for PSTN gateways.  Being that that I-D
> is dealing with PSTN interworking (where the far end typically provides
> audible call progress indications) a convention is needed that addresses
> when local vs. far-end call progress indication is needed.  I'm a little
> behind in my SIP-ISUP reading but I think the precedures described
> therein apply to PSTN gateways and not general SIP UAs.  Also, PSTN
> interworking is a distinct enough scenario that additional semantics
> (for 180 vs. 183 and presence of SDP) are OK (but only as far as
> gateways are concerned).
>
> >
> > > AFAIK, it's not possible in SIP to indicate to a UAC that
> > > it should stop rendering media and go back to local generation of
> > > progress indication.  Nor do we want to try to do this in SIP.
> >
> > [CHH] If that is the case, the text in the drafts I refered
> > to should be
> > rewritten. I guess what you propose is:
> >
> > "When the UAC receives the 180 it does the ringback decission
> > based on if
> > there is an early media session established, and the state of
> > it (sendrecv,
> > inactive,...) - NOT based on if there is a SPD in the 180 or
> > not (of course,
> > the SDP in 180 may be the SDP used to establish the early
> > media session, but
> > that is a separate issue)."
>
> Again, I didn't find this kind of discussion in bis-09.  I myself think
> some guidance (as you suggest above) would be helpful in a some I-D.
> Otherwise we have to debate this aspect on mailing lists :)
>
> >
> > > IMHO, services/applications that will generate early media
> > and then stop
> > > have the responsibility to make the caller's user
> > experience acceptable
> > > (i.e., by playing an announcement, music, or whatever).  In
> > Christer's
> > > B2BUA example, the B2BUA's UAC assumes the responsibility to provide
> > > call progress indication to its user (which is another UAC).
> >
> > [CHH] If that's fine with those who are going to build/use such
> > applications, it's fine with me :) I assumed, however, that
> > people don't
> > want the B2BUA to generate ringback tones on behalf of the
> > SIP phones...
> > And, I also think it could be a relatively common SIP
> > application, so I
> > don't think we're talking about a strange and weird scenarios...
>
> Well, I think the B2BUA has to "manage" the callee in this case, given
> the absence within the protocol to tell a UAC to generate ringback (in
> the general case).  I think it would be a useful capability but doubt
> attaching additional semantics to SDP presence and/or 180/183 is the way
> to do it.  Currently it's only a MAY in bis-09 that a UAC generates
> local ringback upon receipt of a 180 anyway.
>
> >
> > > We should only recommend that a UAC generate a local indication upon
> > > receipt of a 180 and then stop upon receipt of media (which
> > it should
> > > render).  Services such as Christer's example must take it upon
> > > themselves to provide an acceptable user experience.
> > Otherwise we add
> > > complexity to the protocol and degrade both
> > interoperability and user
> > > experience.
> >
> > [CHH] I don't think the UAC shall "stop upon receipt of
> > media". If the UAS
> > starts sending media, it shall also send a 18x with an answer
> > to the offer
> > sent in INVITE to establish an early media session (or, an
> > UPDATE it the
> > early session already exists, but in "inactive" mode).
>
> Again, AFAIK a UAS can send media to the UAC after receiving the UAC's
> SDP.  No response is required to be sent back to the UAC.  Sending a
> response with SDP is the "polite" thing to do IMHO as it's an explicit
> indication that the UAS will send media (and allows RTP/RTCP to "work").
>
> I believe UAC behavior here is an largely an implementation issue, and
> what I said are my views on what will provide an acceptable user
> experience in the general case.  If the SIP WG wants to define
> "operational semantics" for some protocol scenarios I'm OK with that.
> The subject of this thread is a good one IMHO, and I remember a time
> when this was discussed.
>
> >
> > I also would like to get a clarification if the offer/answer
> > rules in bis-09
> > "allow" us to establish an early media session by sending the
> > answer in an
> > unreliable 18x?
>
> My read indicates answers MUST be in a reliable response, which is only
> 200 in bis-09.  This is stated pretty clearly IMHO.  I'm not sure what a
> "early media session" is but if you want to complete the offer-answer
> exchange prior to sending/receiving a 200 OK, then there's the 100-rel
> extension (as I'm sure you're aware :).
>
> Best regards,
> Bert
>
> >
> > > This is what I'd propose be the "official" clarification.
> > My 2 cents
> > > anyway.
> >
> > [CHH] That's what I asked for :)
> >
> > Regards,
> >
> > Christer Holmberg
> > Ericsson Finland
> >
> >
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@ns.ietf.org  Wed Jul  3 14:45:10 2002
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From: "Mahey, Sonit" <Sonit.Mahey@icn.siemens.com>
To: "'Michael Thomas'" <mat@cisco.com>, sip@ietf.org
Subject: RE: [Sip] p-asserted-identity security harmful
Date: Wed, 3 Jul 2002 12:01:46 -0400 
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Folks,

Pardon me for jumping in late in this discussion. I am missing the rationale
behind the choice of TLS as opposed to IPSec.
However, one issue still remains pertinet to both - that of PKI. 
Packet Cable incidently, adopts Kerberos as key management on a few
interfaces. 
First off, is SIP trying to mandate something for security? If yes, why not
adopt IPSec and use Kerberos for Key management, just like PacketCable? 

Thanks.

- sonit

Sonit Mahey,
Systems Architect,
Siemens TGOS,
*: 10 Technology Park Dr., Westford, MA 01886
': 978.923.3271
*: Sonit.Mahey@icn.siemens.com 
http://www.siemens.com

-----Original Message-----
From: Michael Thomas [mailto:mat@cisco.com]
Sent: Wednesday, July 03, 2002 11:41 AM
To: sip@ietf.org
Subject: [Sip] p-asserted-identity security harmful



From Section 11.3 of draft-ietf-sip-asserted-identity-01:

>    Connections between nodes within the Trust Domain and between UAs and
>    nodes in the Trust Domain MUST use TLS using a cipher suite of
>    RSA_WITH_AES_128_CBC_SHA1.  Mutual authentication between nodes in the
>    trust domain MUST be performed and confidentiality MUST be negotiated.
> 


It's just come to my attention that TLS was
mandated in the asserted identity draft. Not only
was TLS mandated, but certain cipher suites and
_AUTHENTICATION_ mechanisms were made necessary
not only to implement, but to deploy. This is
utterly bogus. This also places new requirements
on UA's, which are not part of the base SIP spec.
That too is bogus, and has been slipped in with
no discussion.

First, it ignores the possible use of IPsec as an
alternate mechanism to protect the bits on the
wire, but far more bogus doesn't allow for the
proper latitude with authentication mechanisms
needed in real life. I know that a lot of people
here think that the PKI kool aide is going to
solve world thirst, but reality has shown that to
be a complete myth. SIP should not be mandating
things for *deployment* which are manifestly
undeployable as the only possible mechanism.
Thirdly, it places new requirements on UA's of not
only implementation, but *deployment* as
well. This is extremely harmful.

This section should either be struck in its
entirety, or refer back to the base SIP spec's
proxy requirements (which are likely to be harmful
too, but at least the damage will be limited to a
single document).

	Mike

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@ns.ietf.org  Wed Jul  3 14:48:15 2002
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Subject: [Sip] p-asserted-identity security harmful
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From Section 11.3 of draft-ietf-sip-asserted-identity-01:

>    Connections between nodes within the Trust Domain and between UAs and
>    nodes in the Trust Domain MUST use TLS using a cipher suite of
>    RSA_WITH_AES_128_CBC_SHA1.  Mutual authentication between nodes in the
>    trust domain MUST be performed and confidentiality MUST be negotiated.
> 


It's just come to my attention that TLS was
mandated in the asserted identity draft. Not only
was TLS mandated, but certain cipher suites and
_AUTHENTICATION_ mechanisms were made necessary
not only to implement, but to deploy. This is
utterly bogus. This also places new requirements
on UA's, which are not part of the base SIP spec.
That too is bogus, and has been slipped in with
no discussion.

First, it ignores the possible use of IPsec as an
alternate mechanism to protect the bits on the
wire, but far more bogus doesn't allow for the
proper latitude with authentication mechanisms
needed in real life. I know that a lot of people
here think that the PKI kool aide is going to
solve world thirst, but reality has shown that to
be a complete myth. SIP should not be mandating
things for *deployment* which are manifestly
undeployable as the only possible mechanism.
Thirdly, it places new requirements on UA's of not
only implementation, but *deployment* as
well. This is extremely harmful.

This section should either be struck in its
entirety, or refer back to the base SIP spec's
proxy requirements (which are likely to be harmful
too, but at least the damage will be limited to a
single document).

	Mike

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@ns.ietf.org  Wed Jul  3 14:57:28 2002
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Date: Wed, 03 Jul 2002 11:51:39 -0500
From: Ben Campbell <bcampbell@dynamicsoft.com>
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Subject: [Sip] Re: MESSAGE advertising in REGISTERs
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Certainly it will be needed to make many server related scenarios work. 
It is optional because MESSAGE can also be used in a pure end-to-end 
mode without assuming that such servers exist. For example, a MESSAGE 
UAC may discover the contact for a MESSAGE UAS from presence information 
(i.e. a contact in a presence doc), then send the MESSAGE request 
directly with no intervening servers. In this architecture, there is not 
necessarily a need to advertise support at a registrar.

Jiri Kuthan wrote:
> Can someone tell me why the MESSAGE I-D mandates advertising
> of support in REGISTERs of a MESSAGE-capable UAC only *optionally*?
> 
> Making it optional gains little knowledge to a SIP server -- a server 
> does not know anything it could reliably use in its processing logic.
> 
> Think of a SIP server which acts as a proxy server and message store.
> If it knew, that downstream UAS was not capable of processing MESSAGEs,
> it could immediately store an incoming MESSAGE, send 202 upstream,
> and relay the message to first UAC which registers and advertises support
> for MESSAGE in REGISTER.
> 
> Because the server does not know it, it has to relay unnecessarily 
> a message to an upstream UAS, which will deny it. Then, whenever 
> a REGISTER arrives, the server will try to dump message store to it,
> (hoping a UAC capable of MESSAGE registered) resulting in quite
> uneffective behaviour.
> 
> -Jiri




_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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To: "'Ranjit Avasarala'" <ranjit.avasarala@bplmail.com>,
        Jonathan Rosenberg
	 <jdrosen@dynamicsoft.com>
Cc: sip-ietf <sip@ietf.org>
Subject: RE: [Sip] syntax of NOTIFY
Date: Wed, 3 Jul 2002 13:39:14 -0400 
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Ranjit:

I answered your question already, both on the list and in a mail
sent directly to you. Please read and understand what I sent before
asking your question again.

/a

> -----Original Message-----
> From: Ranjit Avasarala [mailto:ranjit.avasarala@bplmail.com]
> Sent: Wednesday, July 03, 2002 4:08
> To: jdrosen@dynamicsoft.com
> Cc: sip-ietf
> Subject: [Sip] syntax of NOTIFY
> 
> 
> Hi Jonathan
> 
> As per draft the MIME type in NOTIFY should be 
> application/reginfo+xml.
> but in draft on REFER MIME type for NOTIFy is given as 
> message/sipfrag.
> 
> so which one should be used?
> 
> in my case say I want to register a particular event in the 
> server, so can I
> use SUBSCRIBE / NOTIFY pair?
> with just plain mesaage and no xml.
> 
> Thanks
> Ranjit
> 
> 
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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From daemon@ns.ietf.org  Wed Jul  3 15:02:12 2002
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Date: Wed, 03 Jul 2002 13:19:52 -0500
From: Ben Campbell <bcampbell@dynamicsoft.com>
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CC: sip@ietf.org
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Subject: [Sip] Re: MESSAGE advertising in REGISTERs
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What if there is no registrar?

Jiri Kuthan wrote:
> Though advertising MESSAGE capability in REGISTERs does
> not hurt in any scenario, not advertising hurts in some.
> That's why I would much prefer having it as a MUST.
> 
> -Jiri
> 
> At 06:51 PM 7/3/2002, Ben Campbell wrote:
> 
>>Certainly it will be needed to make many server related scenarios work. It is optional because MESSAGE can also be used in a pure end-to-end mode without assuming that such servers exist. For example, a MESSAGE UAC may discover the contact for a MESSAGE UAS from presence information (i.e. a contact in a presence doc), then send the MESSAGE request directly with no intervening servers. In this architecture, there is not necessarily a need to advertise support at a registrar.
>>
>>Jiri Kuthan wrote:
>>
>>>Can someone tell me why the MESSAGE I-D mandates advertising
>>>of support in REGISTERs of a MESSAGE-capable UAC only *optionally*?
>>>Making it optional gains little knowledge to a SIP server -- a server does not know anything it could reliably use in its processing logic.
>>>Think of a SIP server which acts as a proxy server and message store.
>>>If it knew, that downstream UAS was not capable of processing MESSAGEs,
>>>it could immediately store an incoming MESSAGE, send 202 upstream,
>>>and relay the message to first UAC which registers and advertises support
>>>for MESSAGE in REGISTER.
>>>Because the server does not know it, it has to relay unnecessarily a message to an upstream UAS, which will deny it. Then, whenever a REGISTER arrives, the server will try to dump message store to it,
>>>(hoping a UAC capable of MESSAGE registered) resulting in quite
>>>uneffective behaviour.
>>>-Jiri
>>




_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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From daemon@ns.ietf.org  Wed Jul  3 15:07:11 2002
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To: Ben Campbell <bcampbell@dynamicsoft.com>
From: Jiri Kuthan <kuthan@fokus.gmd.de>
Cc: sip@ietf.org
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Though advertising MESSAGE capability in REGISTERs does
not hurt in any scenario, not advertising hurts in some.
That's why I would much prefer having it as a MUST.

-Jiri

At 06:51 PM 7/3/2002, Ben Campbell wrote:
>Certainly it will be needed to make many server related scenarios work. It is optional because MESSAGE can also be used in a pure end-to-end mode without assuming that such servers exist. For example, a MESSAGE UAC may discover the contact for a MESSAGE UAS from presence information (i.e. a contact in a presence doc), then send the MESSAGE request directly with no intervening servers. In this architecture, there is not necessarily a need to advertise support at a registrar.
>
>Jiri Kuthan wrote:
>>Can someone tell me why the MESSAGE I-D mandates advertising
>>of support in REGISTERs of a MESSAGE-capable UAC only *optionally*?
>>Making it optional gains little knowledge to a SIP server -- a server does not know anything it could reliably use in its processing logic.
>>Think of a SIP server which acts as a proxy server and message store.
>>If it knew, that downstream UAS was not capable of processing MESSAGEs,
>>it could immediately store an incoming MESSAGE, send 202 upstream,
>>and relay the message to first UAC which registers and advertises support
>>for MESSAGE in REGISTER.
>>Because the server does not know it, it has to relay unnecessarily a message to an upstream UAS, which will deny it. Then, whenever a REGISTER arrives, the server will try to dump message store to it,
>>(hoping a UAC capable of MESSAGE registered) resulting in quite
>>uneffective behaviour.
>>-Jiri
>


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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--NextPart


A new Request for Comments is now available in online RFC libraries.


        RFC 3265

        Title:	    Session Initiation Protocol (SIP)-Specific Event
                    Notification 
        Author(s):  A. B. Roach
        Status:	    Standards Track
	Date:       June 2002
        Mailbox:    adam@dynamicsoft.com
        Pages:      38
        Characters: 89005
        Obsoletes:  2543

        I-D Tag:    draft-ietf-sip-events-05.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3265.txt


This document describes an extension to the Session Initiation
Protocol (SIP).  The purpose of this extension is to provide an
extensible framework by which SIP nodes can request notification
from remote nodes indicating that certain events have occurred.

Concrete uses of the mechanism described in this document may be
standardized in the future.

Note that the event notification mechanisms defined herein are
NOT intended to be a general-purpose infrastructure for all
classes of event subscription and notification.

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

This is now a Proposed Standard Protocol.

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
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        RFC 3264

        Title:	    An Offer/Answer Model with Session Description
                    Protocol (SDP)
        Author(s):  J. Rosenberg, H. Schulzrinne
        Status:	    Standards Track
	Date:       June 2002
        Mailbox:    jdrosen@dynamicsoft.com,
                    schulzrinne@cs.columbia.edu 
        Pages:      25
        Characters: 60854
        Obsoletes:  2543

        I-D Tag:    draft-ietf-mmusic-sdp-offer-answer-02.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3264.txt


This document defines a mechanism by which two entities can make use
of the Session Description Protocol (SDP) to arrive at a common view
of a multimedia session between them.  In the model, one participant
offers the other a description of the desired session from their
perspective, and the other participant answers with the desired
session from their perspective.  This offer/answer model is most
useful in unicast sessions where information from both participants is
needed for the complete view of the session.  The offer/answer model
is used by protocols like the Session Initiation Protocol (SIP).

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

This is now a Proposed Standard Protocol.

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
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--NextPart


A new Request for Comments is now available in online RFC libraries.


        RFC 3261

        Title:	    SIP: Session Initiation Protocol
        Author(s):  J. Rosenberg, H. Schulzrinne, G. Camarillo,
                    A. Johnston, J. Peterson, R. Sparks, M. Handley,
                    E. Schooler 
        Status:	    Standards Track
	Date:       June 2002
        Mailbox:    jdrosen@dynamicsoft.com,
                    schulzrinne@cs.columbia.edu,
                    Gonzalo.Camarillo@ericsson.com, 
                    alan.johnston@wcom.com, jon.peterson@neustar.com, 
                    rsparks@dynamicsoft.com, mjh@icir.org,
                    schooler@research.att.com 
        Pages:      266
        Characters: 647976
        Obsoletes:  2543

        I-D Tag:    draft-ietf-sip-rfc2543bis-09.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3261.txt


This document describes Session Initiation Protocol (SIP), an
application-layer control (signaling) protocol for creating,
modifying, and terminating sessions with one or more
participants.  These sessions include Internet telephone calls,
multimedia distribution, and multimedia conferences.

SIP invitations used to create sessions carry session descriptions
that allow participants to agree on a set of compatible media types.
SIP makes use of elements called proxy servers to help route requests
to the user's current location, authenticate and authorize users for
services, implement provider call-routing policies, and provide
features to users.  SIP also provides a registration function that
allows users to upload their current locations for use by proxy
servers.  SIP runs on top of several different transport protocols.

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

This is now a Proposed Standard Protocol.

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 list and the RFC-DIST list.
Requests to be added to or deleted from the IETF distribution list
should be sent to IETF-REQUEST@IETF.ORG.  Requests to be
added to or deleted from the RFC-DIST distribution list should
be sent to RFC-DIST-REQUEST@RFC-EDITOR.ORG.

Details on obtaining RFCs via FTP or EMAIL may be obtained by sending
an EMAIL message to rfc-info@RFC-EDITOR.ORG with the message body 
help: ways_to_get_rfcs.  For example:

        To: rfc-info@RFC-EDITOR.ORG
        Subject: getting rfcs

        help: ways_to_get_rfcs

Requests for special distribution should be addressed to either the
author of the RFC in question, or to RFC-Manager@RFC-EDITOR.ORG.  Unless
specifically noted otherwise on the RFC itself, all RFCs are for
unlimited distribution.echo 
Submissions for Requests for Comments should be sent to
RFC-EDITOR@RFC-EDITOR.ORG.  Please consult RFC 2223, Instructions to RFC
Authors, for further information.


Joyce K. Reynolds and Sandy Ginoza
USC/Information Sciences Institute

...

Below is the data which will enable a MIME compliant Mail Reader 
implementation to automatically retrieve the ASCII version
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--NextPart
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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@ns.ietf.org  Wed Jul  3 16:32:50 2002
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To: IETF-Announce: ;
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From: rfc-editor@rfc-editor.org
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Date: Wed, 03 Jul 2002 12:50:01 -0700
Subject: [Sip] RFC 3262 on Reliability of Provisional Responses in the Session Initiation Protocol (SIP)
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        RFC 3262

        Title:	    Reliability of Provisional Responses in Session
                    Initiation Protocol (SIP)
        Author(s):  jdrosen@dynamicsoft.com,
                    schulzrinne@cs.columbia.edu 
        Status:	    Standards Track
	Date:       June 2002
        Mailbox:    jdrosen@dynamicsoft.com,
                    schulzrinne@cs.columbia.edu 
        Pages:      14
        Characters: 29643
        Obsoletes:  2543

        I-D Tag:    draft-ietf-sip-100rel-06.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3262.txt


This document specifies an extension to the Session Initiation
Protocol (SIP) providing reliable provisional response messages.  This
extension uses the option tag 100rel and defines the Provisional
Response ACKnowledgement (PRACK) method.

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

This is now a Proposed Standard Protocol.

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 list and the RFC-DIST list.
Requests to be added to or deleted from the IETF distribution list
should be sent to IETF-REQUEST@IETF.ORG.  Requests to be
added to or deleted from the RFC-DIST distribution list should
be sent to RFC-DIST-REQUEST@RFC-EDITOR.ORG.

Details on obtaining RFCs via FTP or EMAIL may be obtained by sending
an EMAIL message to rfc-info@RFC-EDITOR.ORG with the message body 
help: ways_to_get_rfcs.  For example:

        To: rfc-info@RFC-EDITOR.ORG
        Subject: getting rfcs

        help: ways_to_get_rfcs

Requests for special distribution should be addressed to either the
author of the RFC in question, or to RFC-Manager@RFC-EDITOR.ORG.  Unless
specifically noted otherwise on the RFC itself, all RFCs are for
unlimited distribution.echo 
Submissions for Requests for Comments should be sent to
RFC-EDITOR@RFC-EDITOR.ORG.  Please consult RFC 2223, Instructions to RFC
Authors, for further information.


Joyce K. Reynolds and Sandy Ginoza
USC/Information Sciences Institute

...

Below is the data which will enable a MIME compliant Mail Reader 
implementation to automatically retrieve the ASCII version
of the RFCs.

--NextPart
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--OtherAccess
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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@ns.ietf.org  Wed Jul  3 16:35:09 2002
Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged))
	by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA05486
	for <sip-archive@odin.ietf.org>; Wed, 3 Jul 2002 16:35:09 -0400 (EDT)
Received: (from daemon@localhost)
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	for sip-archive@odin.ietf.org; Wed, 3 Jul 2002 16:35:57 -0400 (EDT)
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	Wed, 3 Jul 2002 14:57:40 +0200
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X-Sender: jiri@iptel.org
X-Mailer: QUALCOMM Windows Eudora Version 5.1
Date: Wed, 03 Jul 2002 14:57:37 +0200
To: sip@ietf.org
From: Jiri Kuthan <jiri@iptel.org>
Cc: bcampbell@dynamicsoft.com
Mime-Version: 1.0
Content-Type: text/plain; charset="us-ascii"
Subject: [Sip] MESSAGE advertising in REGISTERs
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Can someone tell me why the MESSAGE I-D mandates advertising
of support in REGISTERs of a MESSAGE-capable UAC only *optionally*?

Making it optional gains little knowledge to a SIP server -- a server 
does not know anything it could reliably use in its processing logic.

Think of a SIP server which acts as a proxy server and message store.
If it knew, that downstream UAS was not capable of processing MESSAGEs,
it could immediately store an incoming MESSAGE, send 202 upstream,
and relay the message to first UAC which registers and advertises support
for MESSAGE in REGISTER.

Because the server does not know it, it has to relay unnecessarily 
a message to an upstream UAS, which will deny it. Then, whenever 
a REGISTER arrives, the server will try to dump message store to it,
(hoping a UAC capable of MESSAGE registered) resulting in quite
uneffective behaviour.

-Jiri



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@ns.ietf.org  Wed Jul  3 16:35:16 2002
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To: IETF-Announce: ;
Cc: rfc-editor@rfc-editor.org, sip@ietf.org
From: rfc-editor@rfc-editor.org
Mime-Version: 1.0
Content-Type: Multipart/Mixed; Boundary=NextPart
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Subject: [Sip] RFC 3263 on Session Initiation Protocol (SIP): Locating SIP Servers
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--NextPart


A new Request for Comments is now available in online RFC libraries.


        RFC 3263

        Title:	    Session Initiation Protocol (SIP):
                    Locating SIP Servers 
        Author(s):  J. Rosenberg, H. Schulzrinne
        Status:	    Standards Track
	Date:       June 2002
        Mailbox:    jdrosen@dynamicsoft.com,
                    schulzrinne@cs.columbia.edu 
        Pages:      17
        Characters: 42310
        Obsoletes:  2543

        I-D Tag:    draft-ietf-sip-srv-06.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3263.txt


The Session Initiation Protocol (SIP) uses DNS procedures to allow a
client to resolve a SIP Uniform Resource Identifier (URI) into the IP
address, port, and transport protocol of the next hop to contact.  It
also uses DNS to allow a server to send a response to a backup client
if the primary client has failed.  This document describes those DNS
procedures in detail.

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

This is now a Proposed Standard Protocol.

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 list and the RFC-DIST list.
Requests to be added to or deleted from the IETF distribution list
should be sent to IETF-REQUEST@IETF.ORG.  Requests to be
added to or deleted from the RFC-DIST distribution list should
be sent to RFC-DIST-REQUEST@RFC-EDITOR.ORG.

Details on obtaining RFCs via FTP or EMAIL may be obtained by sending
an EMAIL message to rfc-info@RFC-EDITOR.ORG with the message body 
help: ways_to_get_rfcs.  For example:

        To: rfc-info@RFC-EDITOR.ORG
        Subject: getting rfcs

        help: ways_to_get_rfcs

Requests for special distribution should be addressed to either the
author of the RFC in question, or to RFC-Manager@RFC-EDITOR.ORG.  Unless
specifically noted otherwise on the RFC itself, all RFCs are for
unlimited distribution.echo 
Submissions for Requests for Comments should be sent to
RFC-EDITOR@RFC-EDITOR.ORG.  Please consult RFC 2223, Instructions to RFC
Authors, for further information.


Joyce K. Reynolds and Sandy Ginoza
USC/Information Sciences Institute

...

Below is the data which will enable a MIME compliant Mail Reader 
implementation to automatically retrieve the ASCII version
of the RFCs.

--NextPart
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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@ns.ietf.org  Wed Jul  3 16:41:03 2002
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To: IETF-Announce: ;
Cc: rfc-editor@rfc-editor.org, sip@ietf.org
From: rfc-editor@rfc-editor.org
Mime-Version: 1.0
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Subject: [Sip] RFC 3266 on Support for IPv6 in Session Description Protocol
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--NextPart


A new Request for Comments is now available in online RFC libraries.


        RFC 3266

        Title:	    Support for IPv6 in Session Description Protocol
                    (SDP) 
        Author(s):  S. Olson, G. Camarillo, A. B. Roach
        Status:	    Standards Track
	Date:       June 2002
        Mailbox:    seanol@microsoft.com,
                    Gonzalo.Camarillo@ericsson.com,
                    adam@dynamicsoft.com 
        Pages:      5
        Characters: 8693
        Updates:    2327

        I-D Tag:    draft-ietf-mmusic-sdp-ipv6-03.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3266.txt


This document describes the use of Internet Protocol Version 6 (IPv6)
addresses in conjunction with the Session Description Protocol (SDP).
Specifically, this document clarifies existing text in SDP with
regards to the syntax of IPv6 addresses.

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

This is now a Proposed Standard Protocol.

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 list and the RFC-DIST list.
Requests to be added to or deleted from the IETF distribution list
should be sent to IETF-REQUEST@IETF.ORG.  Requests to be
added to or deleted from the RFC-DIST distribution list should
be sent to RFC-DIST-REQUEST@RFC-EDITOR.ORG.

Details on obtaining RFCs via FTP or EMAIL may be obtained by sending
an EMAIL message to rfc-info@RFC-EDITOR.ORG with the message body 
help: ways_to_get_rfcs.  For example:

        To: rfc-info@RFC-EDITOR.ORG
        Subject: getting rfcs

        help: ways_to_get_rfcs

Requests for special distribution should be addressed to either the
author of the RFC in question, or to RFC-Manager@RFC-EDITOR.ORG.  Unless
specifically noted otherwise on the RFC itself, all RFCs are for
unlimited distribution.echo 
Submissions for Requests for Comments should be sent to
RFC-EDITOR@RFC-EDITOR.ORG.  Please consult RFC 2223, Instructions to RFC
Authors, for further information.


Joyce K. Reynolds and Sandy Ginoza
USC/Information Sciences Institute

...

Below is the data which will enable a MIME compliant Mail Reader 
implementation to automatically retrieve the ASCII version
of the RFCs.

--NextPart
Content-Type: Multipart/Alternative; Boundary="OtherAccess"

--OtherAccess
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        access-type="mail-server";
        server="RFC-INFO@RFC-EDITOR.ORG"

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RETRIEVE: rfc
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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@ns.ietf.org  Wed Jul  3 19:07:04 2002
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To: IETF-Announce: ;
Cc: rfc-editor@rfc-editor.org, sip@ietf.org
From: rfc-editor@rfc-editor.org
Mime-Version: 1.0
Content-Type: Multipart/Mixed; Boundary=NextPart
Date: Wed, 03 Jul 2002 13:14:39 -0700
Subject: [Sip] RFC 3267 on Real-Time Transport Protocol (RTP) Payload Format and File Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband (AMR-WB) Audio Codecs
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A new Request for Comments is now available in online RFC libraries.


        RFC 3267

        Title:	    Real-Time Transport Protocol (RTP) Payload Format
                    and File Storage Format for the Adaptive
                    Multi-Rate (AMR) and Adaptive Multi-Rate Wideband
                    (AMR-WB) Audio Codecs 
        Author(s):  J. Sjoberg, M. Westerlund, A. Lakaniemi, Q. Xie
        Status:	    Standards Track
	Date:       June 2002
        Mailbox:    Johan.Sjoberg@ericsson.com,
                    Magnus.Westerlund@ericsson.com,
                    ari.lakaniemi@nokia.com, qxie1@email.mot.com
        Pages:      49
        Characters: 110652
        Updates/Obsoletes/SeeAlso:  None

        I-D Tag:    draft-ietf-avt-rtp-amr-13.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3267.txt


This document specifies a real-time transport protocol (RTP)
payload format to be used for Adaptive Multi-Rate (AMR) and
Adaptive Multi-Rate Wideband (AMR-WB) encoded speech
signals.  The payload format is designed to be able to interoperate
with existing AMR and AMR-WB transport formats on non-IP
networks.  In addition, a file format is specified for transport of
AMR and AMR-WB speech data in storage mode applications such as
email.  Two separate MIME type registrations are included, one for
AMR and one for AMR-WB, specifying use of both the RTP payload
format and the storage format.

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

This is now a Proposed Standard Protocol.

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

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USC/Information Sciences Institute

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From daemon@optimus.ietf.org  Wed Jul  3 20:13:43 2002
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From: Samir Chatterjee <samir.chatterjee@cgu.edu>
To: "'sip@ietf.org'" <sip@ietf.org>
Subject: [Sip] Out of band authentication 
Date: Wed, 3 Jul 2002 16:43:58 -0700 
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Can someone point us to I-Ds, drafts, RFCs or articles that talk about
out-of-band authentication in SIP? I am looking for methods beyond
HTTP-digest style that is recommended. Is there any detail description of
using RADIUS, DIAMETER? Also, is anyone aware of single-sign-on
authentication in SIP with enterprise directories?
Thanks
Samir

Dr. Samir Chatterjee
Associate Professor
School of Information Science
Claremont Graduate University
130 East Ninth Street, Claremont, CA 91711-6190
(Ph) 909-607-4651; (Fax) 909-621-8564
Email:samir.chatterjee@cgu.edu


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From daemon@optimus.ietf.org  Thu Jul  4 04:44:36 2002
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Thread-Topic: [Sip-implementors] List of bugs in bis09
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Subject: [Sip] RE: [Sip-implementors] List of bugs in bis09
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Hi Robert,

Has this been done yet? I would be nice to get a list of changes from bis-09 to RFC3261 (and perhaps all the other new RFCs that had some changes to them in the 48 hour period).

Regards,
Hisham

> -----Original Message-----
> From: ext Robert Sparks [mailto:rsparks@dynamicsoft.com]
> Sent: Thursday, June 06, 2002 9:28 PM
> To: sharad@bhartitelesoft.com
> Cc: sip-implementors@cs.columbia.edu
> Subject: Re: [Sip-implementors] List of bugs in bis09
> 
> 
> Other than the list, no. At least not right now.
> 
> There is an open issues report at 
> http://www.softarmor.com/sipwg/issues, but it doesn't
> display bugs that have been reported and just need to be
> resolved. I'll look into building something that does.
> 
> RjS
> 
> On Thu, 2002-06-06 at 01:39, Sharad Goklani wrote:
> > Hi All,
> > 
> > Does a list of filed bugs/issues in bis09 exist somewhere ?
> > 
> > Thanks,
> > Sharad
> > 
> > _______________________________________________
> > Sip-implementors mailing list
> > Sip-implementors@cs.columbia.edu
> > http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
> 
> 
> _______________________________________________
> Sip-implementors mailing list
> Sip-implementors@cs.columbia.edu
> http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
> 

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From daemon@optimus.ietf.org  Thu Jul  4 08:56:18 2002
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From: "Alexeitsev, D" <D.Alexeitsev@telekom.de>
To: bert.culpepper@intervoice-brite.com
Cc: sip@ietf.org, christer.holmberg@lmf.ericsson.se
Subject: Re: [Sip] Local vs Remote ringback on 180
Date: Thu, 4 Jul 2002 14:17:55 +0200 
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Hello Bert, Christer

Thank you for the clarifications

Some comments inline

Again, AFAIK a UAS can send media to the UAC after receiving the UAC's
SDP.  No response is required to be sent back to the UAC.  Sending a
response with SDP is the "polite" thing to do IMHO as it's an explicit
indication that the UAS will send media (and allows RTP/RTCP to "work").

[DA] That was my understanding as well. I think, that in some scenarios, it is better to wait for the answer before playback for the prevention of receiving media from hosts that are not invited to the session. Even by the cost of user convenience.

My read indicates answers MUST be in a reliable response, which is only
200 in bis-09.  This is stated pretty clearly IMHO.  I'm not sure what a
"early media session" is but if you want to complete the offer-answer
exchange prior to sending/receiving a 200 OK, then there's the 100-rel
extension (as I'm sure you're aware :).

[DA] Is the use of the TCP/SCTP is good enough to say that e2e provisional responses with answer are reliable?

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From daemon@optimus.ietf.org  Thu Jul  4 10:40:06 2002
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Thread-Topic: [SIP, Transport] Does sent on TCP (when msg too big) also occurs on server side?
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Subject: [Sip] [SIP, Transport] Does sent on TCP (when msg too big) also occurs on server side?
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Hello,

I have a question about the fact that we should send a message through TCP if its size is too big for UDP. This is specified for the client side (point 18.1.1), however no comment about it is made for the server side.

Does it means that, if we receive a request through UDP, the response (point 18.2.2) should be done through UDP even if the message is too big for UDP (so ignoring the rule given in 18.1.1). 

I agree that it implies that our response is bigger than the resquest but I think nothing forbide these kind of situations

Thanks for your help.

Regards,

Gilles Compienne.

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From daemon@optimus.ietf.org  Thu Jul  4 10:47:00 2002
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From: "Bert Culpepper" <bert.culpepper@intervoice-brite.com>
To: "'Christer Holmberg'" <christer.holmberg@lmf.ericsson.se>
Cc: "'Alexeitsev, D'" <D.Alexeitsev@telekom.de>, <sip@ietf.org>
Subject: RE: [Sip] Local vs Remote ringback on 180
Date: Thu, 4 Jul 2002 10:18:25 -0400
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Hi Christer,

comments below.

Christer Holmberg wrote:
>
> Hi Bert,
>
> As a conclusion to this discussion my proposal for a
> clarification would be
> that the local ringback generation depends on if a early
> media session has
> been established or not when the 180 arrives.
>
> And, IF a UAS sends early media, and then wants the UAC to
> generate local
> ringback, it first have to put the early media on hold, using
> UPDATE, then
> send 180 (the UAC MAY now generate local ringback, since
> there is no early
> media), and when the called party answers the UAS will first
> send a new
> UPDATE, to un-hold the media, then send 200 for the INVITE. I
> still think
> this process is very messy, and "much pain for little gain", but since
> that's the only way SIP allows us to do it I am not going to
> argue against
> it :)

I agree with you.  But with what's written in bis-09 and other I-Ds
there's no guarantee on what the UAC will do with this sequence of
messages as far as rendering session progress.  I believe the issue here
is how to provide a good user experience and as long as the UAC presents
some indication of what's going on in the signaling the user shouldn't
get confused.

>
> I still would like to get a clarification on the
> "UPDATE-and-buffering-of-200) issue I brought up, though :)
> Personally I see
> no reason why to buffer the 200, but the UAS may have to
> buffer the second
> UPDATE (used to un-hold the media) until it has completed the
> first UPDATE
> (used to put media on hold), which means there is again a
> chance of media
> clipping.

Well, I believe the purpose of UPDATE is to modify session state prior
to the INVITE transaction completing.  Given this, I would say UPDATE
transactions have to complete before the INVITE transaction completes.
And yes, there is a chance of clipping here.  So to prevent this, if the
UAS only indicates that it will stop sending media the UAC will still
render received media if/when it arrives.  Some additional session
update is needed anyway to have the UAC send its media to the new
destination right?  This seems to be a case of 3pcc where we need the
UAC to *not* change its SDP during these transactions.

Regards,
Bert

>
> Now, as Bert mentioned, the UAS may start sending early media without
> indicating it (sending 183 with SDP), and then still send 180
> without SDP
> even if it sends early media, so in that case I guess there
> is a risk the
> local ringback WILL get mixed with the early media (but, we
> can get early
> media mixing in forking scenarios, anyway) - UNLESS the UAC
> is able to do
> some decissions based on detected media (I don't think that
> would work very
> well, but it's up to each implementor to decide for him/herself).
>
> Or, does anyone have another proposal?
>
> Thanks!
>
> Regards,
>
> Christer Holmberg
> Ericsson Finland
>


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Subject: RE: [Sip] Local vs Remote ringback on 180
Date: Thu, 4 Jul 2002 10:42:57 -0400
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Hi,

comments below...

Alexeitsev, D wrote:
>
> Hello Bert, Christer
>
> Thank you for the clarifications
>
> Some comments inline
>
> Again, AFAIK a UAS can send media to the UAC after receiving the UAC's
> SDP.  No response is required to be sent back to the UAC.  Sending a
> response with SDP is the "polite" thing to do IMHO as it's an explicit
> indication that the UAS will send media (and allows RTP/RTCP
> to "work").
>
> [DA] That was my understanding as well. I think, that in some
> scenarios, it is better to wait for the answer before
> playback for the prevention of receiving media from hosts
> that are not invited to the session. Even by the cost of user
> convenience.

Yes, this has been discussed before :)  I think this can be handled for
Christer's scenario if the B2BUA completes the initial INVITE
transaction and uses reINVITE instead of UPDATE to provide the service.
Again, the B2BUA will have to manage the caller's user experience in
this case.

>
> My read indicates answers MUST be in a reliable response,
> which is only
> 200 in bis-09.  This is stated pretty clearly IMHO.  I'm not
> sure what a
> "early media session" is but if you want to complete the offer-answer
> exchange prior to sending/receiving a 200 OK, then there's the 100-rel
> extension (as I'm sure you're aware :).
>
> [DA] Is the use of the TCP/SCTP is good enough to say that
> e2e provisional responses with answer are reliable?

I don't think so for "reliable provisional responses".  Retransmissions
are not needed when using TCP but a PRACK from the UAC is still needed
for "reliable provisional responses".

Regards,
Bert

>
> _______________________________________________
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> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
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From daemon@optimus.ietf.org  Thu Jul  4 11:14:48 2002
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From: "Alexeitsev, D" <D.Alexeitsev@telekom.de>
To: bert.culpepper@intervoice-brite.com
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Subject: RE: [Sip] Local vs Remote ringback on 180
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Hello,

comment inline

> [DA] Is the use of the TCP/SCTP is good enough to say that
> e2e provisional responses with answer are reliable?

I don't think so for "reliable provisional responses".  Retransmissions
are not needed when using TCP but a PRACK from the UAC is still needed
for "reliable provisional responses".

[DA] I mean, that the 100rel is only needed for the UDP. The reliable transport protocols do not need a help from session control protocols.

Greetings,
Denis Alexeitsev



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From daemon@optimus.ietf.org  Thu Jul  4 12:09:55 2002
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 occurs on server side?
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Hi,

The transaction should use the same transport (that's my understanding, at least), which means the response uses the same transport as the response.

It is true that the response may be bigger than the request (eg if the 200 response to a INVITE without SDP), but bis-09 DOES take that into consideration, saying that the maximum size for a request is smaller than the
actual maximum size for a SIP UDP message, so that there is "room" for the response to use UDP too, even if the response is bigger than the request.

IF, however, the response becomes too big for UDP, I guess that COULD be indicated to the UAC in an error response to the request, but I currently don't remember which error code (if such even exists...) should be used to
indicate that. It was discussed once, but you may want to check the archieve for details...

Regards,

Christer Holmberg
Ericsson Finland



Gilles Compienne wrote:

> Hello,
>
> I have a question about the fact that we should send a message through TCP if its size is too big for UDP. This is specified for the client side (point 18.1.1), however no comment about it is made for the server side.
>
> Does it means that, if we receive a request through UDP, the response (point 18.2.2) should be done through UDP even if the message is too big for UDP (so ignoring the rule given in 18.1.1).
>
> I agree that it implies that our response is bigger than the resquest but I think nothing forbide these kind of situations
>
> Thanks for your help.
>
> Regards,
>
> Gilles Compienne.
>
> _______________________________________________
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From daemon@optimus.ietf.org  Thu Jul  4 12:10:07 2002
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To: "'Alexeitsev, D'" <D.Alexeitsev@telekom.de>
Cc: <sip@ietf.org>
Subject: RE: [Sip] Local vs Remote ringback on 180
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Hi Denis,

I don't know that the use of TCP is a substitute for 100rel.  PRACK
provides knowledge that the receiver's UA "processed" the provisional
response.  TCP only guarantees the message was received at the host.  I
may be off-base here, if so others can correct me.

Regards,
Bert

Alexeitsev, D wrote:
>
> Hello,
>
> comment inline
>
> > [DA] Is the use of the TCP/SCTP is good enough to say that
> > e2e provisional responses with answer are reliable?
>
> I don't think so for "reliable provisional responses".
> Retransmissions
> are not needed when using TCP but a PRACK from the UAC is still needed
> for "reliable provisional responses".
>
> [DA] I mean, that the 100rel is only needed for the UDP. The
> reliable transport protocols do not need a help from session
> control protocols.
>
> Greetings,
> Denis Alexeitsev
>
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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From daemon@optimus.ietf.org  Thu Jul  4 12:10:21 2002
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Hi Alexeitsev,

I think I get your point :) I'll try to answer.

Like Bert said, when TCP is used, "reliable provisional response" from a SIP point of view means that the response is allowed to contain an
offer/answer. An "unreliable provisional response", EVEN if sent reliably from a transport layer (TCP) point of view, means that it is NOT
allowed to contain an offer/answer (it MAY contain, but the same SDP has to be sent later anyway).

Now, you could argue that every 18x sent using TCP should always be considered as a "reliable provisional response" from a SIP point of view,
and that PRACK is never needed. Now, that WOULD probably work fine - IF:


1. ...PRACK was only used for what it originally was supposed to be used for - making sure provisional responses are delivered reliably and in
order.

HOWEVER, nowadays PRACK also has defined offer/answer rules, which means we in some cases MUST use it to send SDP (for example if a reliable 18x
contains an offer, the PRACK MUST contain the response, so the PRACK MUST be sent).


2. ...there would never be a TCP/UDP conversion in an intermediate proxy. Now the proxy would have to generate the PRACKs on the hop where UDP
is used.


3. ...the SIP APPLICATION would be transport aware. Currently (as I can remember) only the transaction handler, which needs to know if to
retransmit messages or not, needs to be transport aware.

Now, if you want to argue that it's better to make SIP application transport aware, than sending "useless" PRACKs in the network, and you want
proxies to start generating PRACKs, I do NOT want to take part in that discussion :)



Regards,

Christer Holmberg
Ericsson Finland






"Alexeitsev, D" wrote:

> Hello,
>
> comment inline
>
> > [DA] Is the use of the TCP/SCTP is good enough to say that
> > e2e provisional responses with answer are reliable?
>
> I don't think so for "reliable provisional responses".  Retransmissions
> are not needed when using TCP but a PRACK from the UAC is still needed
> for "reliable provisional responses".
>
> [DA] I mean, that the 100rel is only needed for the UDP. The reliable transport protocols do not need a help from session control protocols.
>
> Greetings,
> Denis Alexeitsev
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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



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Subject: RE: [Sip] [SIP, Transport] Does sent on TCP (when msg too big) also occurs on server side?
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From: "James Undery" <jundery@ubiquity.net>
To: "Christer Holmberg" <christer.holmberg@lmf.ericsson.se>,
        "Gilles Compienne" <gcompienne@ubiquity.net>
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> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@lmf.ericsson.se]

> The transaction should use the same transport (that's my 
> understanding, at least), which means the response uses the 
> same transport as the response.
> 
> It is true that the response may be bigger than the request 
> (eg if the 200 response to a INVITE without SDP), but bis-09 
> DOES take that into consideration, saying that the maximum 
> size for a request is smaller than the
> actual maximum size for a SIP UDP message, so that there is 
> "room" for the response to use UDP too, even if the response 
> is bigger than the request.
> 
> IF, however, the response becomes too big for UDP, I guess 
> that COULD be indicated to the UAC in an error response to 
> the request, but I currently don't remember which error code 
> (if such even exists...) should be used to
> indicate that. It was discussed once, but you may want to 
> check the archieve for details...

The trouble with your suggestion of an error (and it's documented in bug#184) is that you can't always return and error. Admittedly it's far fetched but a 200 response to an INVITE crosses the 64k boundary, how can you send an error without messing up the UAS.

James

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From daemon@optimus.ietf.org  Thu Jul  4 13:36:17 2002
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To: James Undery <jundery@ubiquity.net>
CC: Gilles Compienne <gcompienne@ubiquity.net>, sip@ietf.org
Subject: Re: [Sip] [SIP, Transport] Does sent on TCP (when msg too big) also 
 occurs on server side?
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Hi James,

James Undery wrote:

> > -----Original Message-----
> > From: Christer Holmberg [mailto:christer.holmberg@lmf.ericsson.se]
>
> > The transaction should use the same transport (that's my
> > understanding, at least), which means the response uses the
> > same transport as the response.
> >
> > It is true that the response may be bigger than the request
> > (eg if the 200 response to a INVITE without SDP), but bis-09
> > DOES take that into consideration, saying that the maximum
> > size for a request is smaller than the
> > actual maximum size for a SIP UDP message, so that there is
> > "room" for the response to use UDP too, even if the response
> > is bigger than the request.
> >
> > IF, however, the response becomes too big for UDP, I guess
> > that COULD be indicated to the UAC in an error response to
> > the request, but I currently don't remember which error code
> > (if such even exists...) should be used to
> > indicate that. It was discussed once, but you may want to
> > check the archieve for details...
>
> The trouble with your suggestion of an error (and it's documented in bug#184) is that you can't always return and error. Admittedly it's far fetched but a 200 response to an INVITE crosses the 64k boundary, how can you send an error without messing up the UAS.

It's not a suggestion - it's the closest to a suggestion I can think of :) As you said yourself, it's an existing and unfixed bug, so I guess there really is no good solution at the moment.

The problem, of course, is that if the UAS sends a 4xx response it will most likely also release the call it just established (I assume that's what you meant by saying that "it will mess up the UAS"). That, of course, is stupid since the called party just has
picked up the hook (the reason why the 200 is sent), and is most likely going to receive a NEW INVITE (now using TCP) very soon anyway.

The best thing would be if the UAS can figure out how big the 200 is going to be before it even alerts the called party, or at least before the called party picks up the phone. In a "manyfolks" scenario I guess that will work, but probably not as well in cases
where the called party is alerted as soon as the INVITE is received...

Anyway, as you say, it's "far fetched", and I guess the assumption is that a 200 will never be too big. But, never forget Murphy's law... :)

Regards,

Christer Holmberg
Ericsson Finland



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From daemon@optimus.ietf.org  Thu Jul  4 15:52:05 2002
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From: "Peterson, Jon" <jon.peterson@neustar.biz>
To: "'Michael Thomas'" <mat@cisco.com>, sip@ietf.org
Subject: RE: [Sip] p-asserted-identity security harmful
Date: Thu, 4 Jul 2002 14:27:24 -0500 
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Um, did you perhaps not see the word 'Example' in the title of section 11?
The document only requires that a Spec(T) be defined, and provides one in
section 11 for illustrative purposes. Or as it says in the beginning of
section 11:

   The remainder of this section presents an example Spec(T), which is
   not normative in any way.

Thanks as always for your reasoned analysis of our work in this group.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Michael Thomas [mailto:mat@cisco.com]
> Sent: Wednesday, July 03, 2002 8:41 AM
> To: sip@ietf.org
> Subject: [Sip] p-asserted-identity security harmful
> 
> 
> 
> From Section 11.3 of draft-ietf-sip-asserted-identity-01:
> 
> >    Connections between nodes within the Trust Domain and 
> between UAs and
> >    nodes in the Trust Domain MUST use TLS using a cipher suite of
> >    RSA_WITH_AES_128_CBC_SHA1.  Mutual authentication 
> between nodes in the
> >    trust domain MUST be performed and confidentiality MUST 
> be negotiated.
> > 
> 
> 
> It's just come to my attention that TLS was
> mandated in the asserted identity draft. Not only
> was TLS mandated, but certain cipher suites and
> _AUTHENTICATION_ mechanisms were made necessary
> not only to implement, but to deploy. This is
> utterly bogus. This also places new requirements
> on UA's, which are not part of the base SIP spec.
> That too is bogus, and has been slipped in with
> no discussion.
> 
> First, it ignores the possible use of IPsec as an
> alternate mechanism to protect the bits on the
> wire, but far more bogus doesn't allow for the
> proper latitude with authentication mechanisms
> needed in real life. I know that a lot of people
> here think that the PKI kool aide is going to
> solve world thirst, but reality has shown that to
> be a complete myth. SIP should not be mandating
> things for *deployment* which are manifestly
> undeployable as the only possible mechanism.
> Thirdly, it places new requirements on UA's of not
> only implementation, but *deployment* as
> well. This is extremely harmful.
> 
> This section should either be struck in its
> entirety, or refer back to the base SIP spec's
> proxy requirements (which are likely to be harmful
> too, but at least the damage will be limited to a
> single document).
> 
> 	Mike
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From daemon@optimus.ietf.org  Thu Jul  4 16:01:25 2002
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To: "Peterson, Jon" <jon.peterson@neustar.biz>
Cc: "'Michael Thomas'" <mat@cisco.com>, sip@ietf.org
Subject: RE: [Sip] p-asserted-identity security harmful
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Jon,

I am not the only one who was confused by this and
before you start copping attitude you might
consider whether REQUIREMENTS LANGUAGE in EXAMPLES
is a recipe for an understandable document.

	    Mike

Peterson, Jon writes:
 > Um, did you perhaps not see the word 'Example' in the title of section 11?
 > The document only requires that a Spec(T) be defined, and provides one in
 > section 11 for illustrative purposes. Or as it says in the beginning of
 > section 11:
 > 
 >    The remainder of this section presents an example Spec(T), which is
 >    not normative in any way.
 > 
 > Thanks as always for your reasoned analysis of our work in this group.
 > 
 > Jon Peterson
 > NeuStar, Inc.
 > 
 > > -----Original Message-----
 > > From: Michael Thomas [mailto:mat@cisco.com]
 > > Sent: Wednesday, July 03, 2002 8:41 AM
 > > To: sip@ietf.org
 > > Subject: [Sip] p-asserted-identity security harmful
 > > 
 > > 
 > > 
 > > From Section 11.3 of draft-ietf-sip-asserted-identity-01:
 > > 
 > > >    Connections between nodes within the Trust Domain and 
 > > between UAs and
 > > >    nodes in the Trust Domain MUST use TLS using a cipher suite of
 > > >    RSA_WITH_AES_128_CBC_SHA1.  Mutual authentication 
 > > between nodes in the
 > > >    trust domain MUST be performed and confidentiality MUST 
 > > be negotiated.
 > > > 
 > > 
 > > 
 > > It's just come to my attention that TLS was
 > > mandated in the asserted identity draft. Not only
 > > was TLS mandated, but certain cipher suites and
 > > _AUTHENTICATION_ mechanisms were made necessary
 > > not only to implement, but to deploy. This is
 > > utterly bogus. This also places new requirements
 > > on UA's, which are not part of the base SIP spec.
 > > That too is bogus, and has been slipped in with
 > > no discussion.
 > > 
 > > First, it ignores the possible use of IPsec as an
 > > alternate mechanism to protect the bits on the
 > > wire, but far more bogus doesn't allow for the
 > > proper latitude with authentication mechanisms
 > > needed in real life. I know that a lot of people
 > > here think that the PKI kool aide is going to
 > > solve world thirst, but reality has shown that to
 > > be a complete myth. SIP should not be mandating
 > > things for *deployment* which are manifestly
 > > undeployable as the only possible mechanism.
 > > Thirdly, it places new requirements on UA's of not
 > > only implementation, but *deployment* as
 > > well. This is extremely harmful.
 > > 
 > > This section should either be struck in its
 > > entirety, or refer back to the base SIP spec's
 > > proxy requirements (which are likely to be harmful
 > > too, but at least the damage will be limited to a
 > > single document).
 > > 
 > > 	Mike
 > > 
 > > _______________________________________________
 > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
 > > This list is for NEW development of the core SIP Protocol
 > > Use sip-implementors@cs.columbia.edu for questions on current sip
 > > Use sipping@ietf.org for new developments on the application of sip
 > > 
 > 
 > _______________________________________________
 > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
 > This list is for NEW development of the core SIP Protocol
 > Use sip-implementors@cs.columbia.edu for questions on current sip
 > Use sipping@ietf.org for new developments on the application of sip

_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Thu Jul  4 16:32:47 2002
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You are, of course, correct that requirements language in examples is 
odd and potentially confusing. However, spec(T), of which this is an 
example, is supposed to be a specification itself - defining exactly 
what it means for an identity to be asserted.

-Jonathan R.

Michael Thomas wrote:
> Jon,
> 
> I am not the only one who was confused by this and
> before you start copping attitude you might
> consider whether REQUIREMENTS LANGUAGE in EXAMPLES
> is a recipe for an understandable document.
> 
> 	    Mike
> 
> Peterson, Jon writes:
>  > Um, did you perhaps not see the word 'Example' in the title of
> section 11?
>  > The document only requires that a Spec(T) be defined, and provides
> one in
>  > section 11 for illustrative purposes. Or as it says in the beginning
> of
>  > section 11:
>  > 
>  >    The remainder of this section presents an example Spec(T), which
> is
>  >    not normative in any way.
>  > 
>  > Thanks as always for your reasoned analysis of our work in this
> group.
>  > 
>  > Jon Peterson
>  > NeuStar, Inc.
>  > 
>  > > -----Original Message-----
>  > > From: Michael Thomas [mailto:mat@cisco.com]
>  > > Sent: Wednesday, July 03, 2002 8:41 AM
>  > > To: sip@ietf.org
>  > > Subject: [Sip] p-asserted-identity security harmful
>  > > 
>  > > 
>  > > 
>  > > From Section 11.3 of draft-ietf-sip-asserted-identity-01:
>  > > 
>  > > >    Connections between nodes within the Trust Domain and 
>  > > between UAs and
>  > > >    nodes in the Trust Domain MUST use TLS using a cipher suite of
>  > > >    RSA_WITH_AES_128_CBC_SHA1.  Mutual authentication 
>  > > between nodes in the
>  > > >    trust domain MUST be performed and confidentiality MUST 
>  > > be negotiated.
>  > > > 
>  > > 
>  > > 
>  > > It's just come to my attention that TLS was
>  > > mandated in the asserted identity draft. Not only
>  > > was TLS mandated, but certain cipher suites and
>  > > _AUTHENTICATION_ mechanisms were made necessary
>  > > not only to implement, but to deploy. This is
>  > > utterly bogus. This also places new requirements
>  > > on UA's, which are not part of the base SIP spec.
>  > > That too is bogus, and has been slipped in with
>  > > no discussion.
>  > > 
>  > > First, it ignores the possible use of IPsec as an
>  > > alternate mechanism to protect the bits on the
>  > > wire, but far more bogus doesn't allow for the
>  > > proper latitude with authentication mechanisms
>  > > needed in real life. I know that a lot of people
>  > > here think that the PKI kool aide is going to
>  > > solve world thirst, but reality has shown that to
>  > > be a complete myth. SIP should not be mandating
>  > > things for *deployment* which are manifestly
>  > > undeployable as the only possible mechanism.
>  > > Thirdly, it places new requirements on UA's of not
>  > > only implementation, but *deployment* as
>  > > well. This is extremely harmful.
>  > > 
>  > > This section should either be struck in its
>  > > entirety, or refer back to the base SIP spec's
>  > > proxy requirements (which are likely to be harmful
>  > > too, but at least the damage will be limited to a
>  > > single document).
>  > > 
>  > > 	Mike
>  > > 
>  > > _______________________________________________
>  > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>  > > This list is for NEW development of the core SIP Protocol
>  > > Use sip-implementors@cs.columbia.edu for questions on current sip
>  > > Use sipping@ietf.org for new developments on the application of sip
>  > > 
>  > 
>  > _______________________________________________
>  > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>  > This list is for NEW development of the core SIP Protocol
>  > Use sip-implementors@cs.columbia.edu for questions on current sip
>  > Use sipping@ietf.org for new developments on the application of sip
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 


-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Thu Jul  4 16:46:54 2002
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Date: Thu, 04 Jul 2002 16:17:01 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: Jiri Kuthan <jiri@iptel.org>
CC: sip@ietf.org, bcampbell@dynamicsoft.com
Subject: Re: [Sip] MESSAGE advertising in REGISTERs
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Jiri Kuthan wrote:
> Can someone tell me why the MESSAGE I-D mandates advertising
> of support in REGISTERs of a MESSAGE-capable UAC only *optionally*?

Because the system will function without a proxy knowing this ahead of 
time. To date, we have not mandated the use of caller preferences (which 
is where this comes from) in other sip extensions, such as UPDATE, 
PRACK. I think presence mentions that you SHOULD do this, but not RFC 
3265 (i.e.,  it doesnt apply to all events).

Indeed, since not all methods are mandatory even in baseline SIP, and 
indeed, rfc3261 doesn't actually mandate anything but OPTIONS, we 
already have the case that a proxy doesn't know the methods of the UAs 
that register to it.

> 
> Making it optional gains little knowledge to a SIP server -- a server 
> does not know anything it could reliably use in its processing logic.

Sure it does. If it is there, it knows that the UA does or does not 
support MESSAGE. Its just that its not always there.

> 
> Think of a SIP server which acts as a proxy server and message store.
> If it knew, that downstream UAS was not capable of processing MESSAGEs,
> it could immediately store an incoming MESSAGE, send 202 upstream,
> and relay the message to first UAC which registers and advertises support
> for MESSAGE in REGISTER.
> 
> Because the server does not know it, it has to relay unnecessarily 
> a message to an upstream UAS, which will deny it. 

That doesn't seem so bad.

If it bothers you, send an OPTIONS to the UA once it registers.



> Then, whenever 
> a REGISTER arrives, the server will try to dump message store to it,
> (hoping a UAC capable of MESSAGE registered) resulting in quite
> uneffective behaviour.

Well, that is only one way to handle transmission of undelivered 
messages to a UA when it comes online. For many reasons, it is not the 
best way, I think. A better way is allow a UA to query to see if it has 
any undelivered messages when it registers, and then pull them as 
needed. This works better in wireless, too. Even if the UA does support 
MESSAGE, it may not want all of them the moment it connects.


-Jonathan R.



-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Thu Jul  4 16:48:19 2002
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From: "Prabhas Sinha" <prsinha@unity.ncsu.edu>
To: <sip@ietf.org>
Date: Thu, 4 Jul 2002 16:21:43 -0400
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Subject: [Sip] Rewording of definitions in Section 6 of RFC 2543-bis09
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This is a multi-part message in MIME format.

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Hi Johnathan,
These are the following things which I feel should be reworded for =
better understanding:

 Section 6, rfc2543bis-09

1. definition of B2BUA on line 610 and definition of UA on line 753 =
doesnt reflect any differences between the two.

2. definition of "Call Stateful" on line 618 and  "Stateful Proxy" on =
line 720 also doesnt differentiate between the two.

3. definition of "Loose Routing" on line 658 and "Strict Routing" on =
line 727 also doesnt give a clear picture of the difference between the =
two.

a good example good be the way "Loop" on line 653 and "Spiral" on line =
713 has been defined. It clearly shows the difference between the two.

Please let me hear what the WG feels about it.

Thanks,

Prabhas

------=_NextPart_000_27AE_01C22376.E278DD00
Content-Type: text/html;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2713.1100" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hi Johnathan,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>These are the following things which I =
feel should=20
be reworded for better understanding:</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;Section 6, =
rfc2543bis-09</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>1. definition of B2BUA on line 610 and =
definition=20
of UA on line 753 doesnt reflect any differences between the =
two.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>2. definition of "Call Stateful" on =
line 618=20
and&nbsp; "Stateful Proxy" on line 720 also doesnt differentiate between =
the=20
two.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>3. definition of "Loose Routing" on =
line 658 and=20
"Strict Routing" on line 727 also doesnt give a clear picture of the =
difference=20
between the two.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>a good example good be the way "Loop" =
on line 653=20
and "Spiral" on line 713 has been defined. It clearly shows the =
difference=20
between the two.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Please let me hear what the WG feels =
about=20
it.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Thanks,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Prabhas</FONT></DIV></BODY></HTML>

------=_NextPart_000_27AE_01C22376.E278DD00--


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From daemon@optimus.ietf.org  Fri Jul  5 06:11:16 2002
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From: Euchner Martin  ICN M SR 3 <Martin.Euchner@icn.siemens.de>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
        Michael Thomas
	 <mat@cisco.com>,
        Euchner Martin  ICN M SR 3
	 <Martin.Euchner@icn.siemens.de>
Cc: "Peterson, Jon" <jon.peterson@neustar.biz>, sip@ietf.org
Subject: RE: [Sip] p-asserted-identity security harmful
Date: Fri, 5 Jul 2002 11:44:54 +0200 
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Now I'm confused even more as this sounds like the Spec(T) example is in
fact of more mandatory nature.

11.2 Authentication requirements
   Users MUST be authenticated using SIP Digest Authentication.

First of all, I do not see any reason why to mandate SIP digest
Authentication for this draft.

Doing that would tie this draft forever to digest, but would not allow any
other SIP authentication mechanism. I recommend to reword as a SHOULD at
least and offering a MAY for other (SIP-09) authentication methods.
I guess what is perhaps intended is simply that "Users MUST/SHOULD be
authenticated".

My understanding of this draft is, that the draft assumes that users have
been authenticated (although even this can be weak in a trusted domain). The
draft describes a means how to populate the privacy-ID information but
certainly does not describe how to secure SIP.

11.3 Security requirements
   Connections between nodes within the Trust Domain and between UAs and
   nodes in the Trust Domain MUST use TLS using a cipher suite of
   RSA_WITH_AES_128_CBC_SHA1.  Mutual authentication between nodes in
   the trust domain MUST be performed and confidentiality MUST be
   negotiated.

Further, I really do not understand and do not see any value why this drafts
actually profiles TLS cipher suites allowing only RSA_AES128, but does not
allow any other defined TLS cipher suites. The defined TLS cipher suites
have been chosen with care and for good reasons; TLS has its inherent
security capability negotiation, that works just fine.

Then, as pointed out by Michael, the MUST mandate for mutual TLS is
counterproductive: if there is mutual TLS authentication, then digest
authentication is completely redundant. On the other hand, it >>should<< be
allowed to have proxy-sided TLS authentication combined with some other
authentication, and of course other authentication methods should not be
ruled out.

Finally, I completely agree with Michael's proposal, that sections 11.2 and
11.3 be completely revised, making them aligned with what SIP-09 spec says
about authentication and security (although that is problematic as well) and
leaving sufficient degree of freedom for future extensions. This privacy
draft shall not attempt to define its own SIP Security baseline.


With kind regards

Martin Euchner.
-----------------------------------------------------------------------
| Dipl.-Inf.                     Rapporteur Q.G/SG16
| Martin Euchner                 Phone: +49 89 722 55790
| Siemens AG.....................Fax  : +49 89 722 47713
| ICN M SR 3                     mailto:Martin.Euchner@icn.siemens.de
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| __________________
| Germany     
-----------------------------------------------------------------------

 -----Original Message-----
From: 	Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com] 
Sent:	Thursday, July 04, 2002 10:03 PM
To:	Michael Thomas
Cc:	Peterson, Jon; sip@ietf.org
Subject:	Re: [Sip] p-asserted-identity security harmful

You are, of course, correct that requirements language in examples is 
odd and potentially confusing. However, spec(T), of which this is an 
example, is supposed to be a specification itself - defining exactly 
what it means for an identity to be asserted.

-Jonathan R.

Michael Thomas wrote:
> Jon,
> 
> I am not the only one who was confused by this and
> before you start copping attitude you might
> consider whether REQUIREMENTS LANGUAGE in EXAMPLES
> is a recipe for an understandable document.
> 
> 	    Mike
> 
> Peterson, Jon writes:
>  > Um, did you perhaps not see the word 'Example' in the title of
> section 11?
>  > The document only requires that a Spec(T) be defined, and provides
> one in
>  > section 11 for illustrative purposes. Or as it says in the beginning
> of
>  > section 11:
>  > 
>  >    The remainder of this section presents an example Spec(T), which
> is
>  >    not normative in any way.
>  > 
>  > Thanks as always for your reasoned analysis of our work in this
> group.
>  > 
>  > Jon Peterson
>  > NeuStar, Inc.
>  > 
>  > > -----Original Message-----
>  > > From: Michael Thomas [mailto:mat@cisco.com]
>  > > Sent: Wednesday, July 03, 2002 8:41 AM
>  > > To: sip@ietf.org
>  > > Subject: [Sip] p-asserted-identity security harmful
>  > > 
>  > > 
>  > > 
>  > > From Section 11.3 of draft-ietf-sip-asserted-identity-01:
>  > > 
>  > > >    Connections between nodes within the Trust Domain and 
>  > > between UAs and
>  > > >    nodes in the Trust Domain MUST use TLS using a cipher suite of
>  > > >    RSA_WITH_AES_128_CBC_SHA1.  Mutual authentication 
>  > > between nodes in the
>  > > >    trust domain MUST be performed and confidentiality MUST 
>  > > be negotiated.
>  > > > 
>  > > 
>  > > 
>  > > It's just come to my attention that TLS was
>  > > mandated in the asserted identity draft. Not only
>  > > was TLS mandated, but certain cipher suites and
>  > > _AUTHENTICATION_ mechanisms were made necessary
>  > > not only to implement, but to deploy. This is
>  > > utterly bogus. This also places new requirements
>  > > on UA's, which are not part of the base SIP spec.
>  > > That too is bogus, and has been slipped in with
>  > > no discussion.
>  > > 
>  > > First, it ignores the possible use of IPsec as an
>  > > alternate mechanism to protect the bits on the
>  > > wire, but far more bogus doesn't allow for the
>  > > proper latitude with authentication mechanisms
>  > > needed in real life. I know that a lot of people
>  > > here think that the PKI kool aide is going to
>  > > solve world thirst, but reality has shown that to
>  > > be a complete myth. SIP should not be mandating
>  > > things for *deployment* which are manifestly
>  > > undeployable as the only possible mechanism.
>  > > Thirdly, it places new requirements on UA's of not
>  > > only implementation, but *deployment* as
>  > > well. This is extremely harmful.
>  > > 
>  > > This section should either be struck in its
>  > > entirety, or refer back to the base SIP spec's
>  > > proxy requirements (which are likely to be harmful
>  > > too, but at least the damage will be limited to a
>  > > single document).
>  > > 
>  > > 	Mike
>  > > 
>  > > _______________________________________________
>  > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>  > > This list is for NEW development of the core SIP Protocol
>  > > Use sip-implementors@cs.columbia.edu for questions on current sip
>  > > Use sipping@ietf.org for new developments on the application of sip
>  > > 
>  > 
>  > _______________________________________________
>  > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>  > This list is for NEW development of the core SIP Protocol
>  > Use sip-implementors@cs.columbia.edu for questions on current sip
>  > Use sipping@ietf.org for new developments on the application of sip
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 


-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com


_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Fri Jul  5 10:51:11 2002
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Message-ID: <3D25AA8F.8040908@acm.org>
Date: Fri, 05 Jul 2002 07:17:51 -0700
From: Marc Petit-Huguenin <petithug@acm.org>
Organization: 8x8, Inc.
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As a SIP implementor, I would like to thanks and congratulate the authors and 
the WG for their work on the RFC326[1-6].

-- 
Marc Petit-Huguenin
Home: mailto:marc@petit-huguenin.org
Work: mailto:petithug@8x8.com
"So much to do and so little time" The Joker - Batman (The Movie)


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From daemon@optimus.ietf.org  Fri Jul  5 14:49:08 2002
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Date: Fri, 05 Jul 2002 14:22:21 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Folks,

I posted an update to draft-ietf-sip-nat. Until it appears in the 
archives, you can pick up a copy at:

http://www.jdrosen.net/papers/draft-ietf-sip-nat-02.txt

As agreed on the list, the Translate header is gone. Its just rport now, 
resulting in a very short and suite specification. The plan is for 
connection reuse to fill the role of the Translate header.

Thanks,
Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Fri Jul  5 14:52:39 2002
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Folks,

I posted an update to the session timer specification before the I-D 
deadline. It should appear in the archives shortly. Until it does, you 
can grab it at:

http://www.jdrosen.net/papers/draft-ietf-sip-session-timer-09.txt

Changes since -08:

* added UPDATE as a valid method to use session timer.

* changed default reason phrase of 422 to "Session Interval too Small"
as part of an improvement in consistency of terminology.

* introduced the term "session refresh request" to refer to INVITE or
UPDATE.

* deleted repetitive normative text in the sections defining the
Min-SE and Session-Expires header fields.

* General update for bis, including grammar, Table 2/3 structure,
terminology.

* removed text on mandatory inclusion of from tags, and on rejecting
requests that are received with tags but that don't match a
dialog. All of that is in bis.

* Proxies used to be able to add a refresher parameter with value uas
into a Session-Expires header field, if the header field was not
present and was added by the proxy. This is redundant, and was the
only case where the proxy could set the refresher parameter. Instead
of this useless exception, the specification now makes a blanket
statement that proxies can't insert or modify the refresher parameter.

* added IANA considerations

* removed old examples section, per sip-guidelines, included a new one
with a complete ladder diagram, full messages, and reduced the number
of different example flows.


This document is now in excellent shape, I believe. I really went 
through it with a fine tooth comb to make it consistent. I do not think 
the above changes (the only significant one was usage of UPDATE) 
warrants a new WGLC, since the change was agreed on the list.

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Fri Jul  5 15:05:17 2002
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Organization: dynamicsoft
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Folks,

I submitted an update to the caller preferences specification. It should 
appear in the archives shortly. Until it does, you can pick up a copy at:

http://www.jdrosen.net/papers/draft-ietf-sip-callerprefs-06.txt

The changes are substantial:

* complete rewrite to be based on rfc2533. It amazes me how much of
the work we did was already done. Fortunately, this results primarily
in a change in presentation, and less of a change in the syntax and
overall behavior from a black box perspective.

* The media contact parameter is now only used to indicate top level
MIME types (audio, video, message), and not full MIME
types. Indicating full MIME types is done through the existing "type"
feature tag, defined in RFC 2913. THis will break interop with people
using the media contact parameter.

* Added new "automata" and "events" caller preferences tags.

* As a consequence of using an ordered token to represent priorities,
we no longer have the ability to handle unknown priorities as equal to
normal (I don't think). In addition, a UA can now register multiple 
priorities,
although this is redundant, since the matching operation is the same
as before - the call will be routed to a UA which has indicated it
supports a priority equal to, or higher than, the priority in the
Priority header of the INVITE.

* The q value has to appear as the last parameter in the
Accept-Contact or Reject-Contact header fields.

* The Accept-COntact and Reject-Contact header field can contain
explicit preferences for methods and events, even though a proxy
computes implicit preferences based on the request (based on the
request method and Event header field value, respectively). Explicit
preferences always override implicitly derived ones.

* Providing preferences for URI are still allowed. However, since they
are done as part of the RFC 2533 matching procedure, the scope is
restricted. Matches are only done on the user part and domain
part. The matching rule is case sensitive match for both. The ability
to match on the user part only, and not the domain, has been removed.

* The specification used to differentiate between the features
expressible in a registration, and those expressable in an
Accept-Contact and Reject-Contact. All of those restrictions are
removed. Both express feature sets, and can contain the same
things. As an example, the spec used to forbid a Contact header in a
REGISTER like:

Contact: sip:1.2.3.4;mobility="!fixed"

since it had a negation. This restriction is removed.

* Eliminated the use of the & construct in the Accept-Contact and
Reject-Contact header fields. This signficantly changes the scope of
feature predicates that can be constructed. An open issue is whether
to bring it back by simply allowing for an actual rfc2533 expression
in the Accept-Contact header.

There are a number of open issues in the spec, all of which are called 
out. Comments are appreciated on them.

Having done the rewrite based on 2533, I am quite happy it was done. Its 
a significant improvement in the specification. The document is much 
shorter - don't let the page count fool you; much of it is IANA 
registrations.

Thanks,
Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Fri Jul  5 16:13:28 2002
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Message-ID: <3D25F75A.9060504@dynamicsoft.com>
Date: Fri, 05 Jul 2002 15:45:30 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: Euchner Martin ICN M SR 3 <Martin.Euchner@icn.siemens.de>
CC: Michael Thomas <mat@cisco.com>,
        "Peterson, Jon"
 <jon.peterson@neustar.biz>, sip@ietf.org
Subject: Re: [Sip] p-asserted-identity security harmful
References: <DA6599EBFD6CF042B0B77964CFF6209553033B@MCHH162E>
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Euchner Martin ICN M SR 3 wrote:
> Now I'm confused even more as this sounds like the Spec(T) example is in
> fact of more mandatory nature.

No, no, no.

p-asserted-identity basically has proxies "declaring" the identity of 
the originator of the request. In order for a proxy to determine what 
this means, each domain has to define, FOR THAT DOMAIN, how the identity 
was determined, and how its securely transmitted between proxies. This 
is not a written specification, or an IETF specification. Its more of an 
enterprise policy. A proxy in a domain has to know what the enterprise 
policy for that domain is, in order to know how to properly interpret 
the asserited identity. Spec(T) is the enterprise policy. The 
specification gives an example of such a policy.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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From daemon@optimus.ietf.org  Fri Jul  5 17:23:37 2002
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To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
        Euchner Martin ICN M SR 3 <Martin.Euchner@icn.siemens.de>
Cc: Michael Thomas <mat@cisco.com>, sip@ietf.org
Subject: RE: [Sip] p-asserted-identity security harmful
Date: Fri, 5 Jul 2002 15:57:46 -0500 
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Let me add to Jonathan's comments that the asserted-identity draft will be
an Informational RFC that declares only P- headers (as specified by the
sipchange process), and as such it is not empowered to make any kind of
normative amendments to RFC3261.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Friday, July 05, 2002 12:46 PM
> To: Euchner Martin ICN M SR 3
> Cc: Michael Thomas; Peterson, Jon; sip@ietf.org
> Subject: Re: [Sip] p-asserted-identity security harmful
> 
> 
> 
> 
> Euchner Martin ICN M SR 3 wrote:
> > Now I'm confused even more as this sounds like the Spec(T) 
> example is in
> > fact of more mandatory nature.
> 
> No, no, no.
> 
> p-asserted-identity basically has proxies "declaring" the identity of 
> the originator of the request. In order for a proxy to determine what 
> this means, each domain has to define, FOR THAT DOMAIN, how 
> the identity 
> was determined, and how its securely transmitted between 
> proxies. This 
> is not a written specification, or an IETF specification. Its 
> more of an 
> enterprise policy. A proxy in a domain has to know what the 
> enterprise 
> policy for that domain is, in order to know how to properly interpret 
> the asserited identity. Spec(T) is the enterprise policy. The 
> specification gives an example of such a policy.
> 
> -Jonathan R.
> 
> 
> -- 
> Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
> Chief Scientist                         First Floor
> dynamicsoft                             East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
> http://www.jdrosen.net                  PH:  (973) 952-5000
> http://www.dynamicsoft.com
> 

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From daemon@optimus.ietf.org  Sat Jul  6 01:10:05 2002
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the sip draft contains a number of examples of calls towards the 
begining and the end. in these, the ip addresses and the domain 
names are not clearly associated with systems.

1) in the figure 1 example, alice's pc is not clearly marked as 
pc33.altanta.com, nor is the atlanta.com proxy labelled as 
atlanta.com (a purist might consider that the proxy is named 
sip.atlanta.com). further down, the hosts 
bibbox3.site3.atlanta.com and server10.biloxi.com are not marked 
in the figure 1 trapezium either. it might be a good idea to write 
the ip addresses and the FQDN of all the hosts in the network 
diagram.

this is of special relevance because the basic fields like From:, 
To:, Via:, Contact: and SIP-URL are often confusing to a novice.

2) Section 26.2 gives a number of examples. the ip addresses in 
these are given without any reference to which hosts they refer to 
in the figure.

- farhan
_________________________________________________________
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From daemon@optimus.ietf.org  Sat Jul  6 11:12:56 2002
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Date: Sat, 06 Jul 2002 16:51:11 +0200
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>, Jiri Kuthan <jiri@iptel.org>
From: Jiri Kuthan <kuthan@fokus.gmd.de>
Subject: Re: [Sip] MESSAGE advertising in REGISTERs
Cc: sip@ietf.org, bcampbell@dynamicsoft.com
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I still don't understand why you're opposed to mandatory 
advertising MESSAGE support in REGISTERs -- I'm not aware
of a harm and I am aware of a benefit in some scenarios.
Sending a message to a destination which could be known from 
its ads not to accept it is simply an unnecessary inefficiency.

Of course, I'm not worried about scenarios to which you 
refered and in which this capability is useless.

-Jiri

At 10:17 PM 7/4/2002, Jonathan Rosenberg wrote:


>Jiri Kuthan wrote:
>>Can someone tell me why the MESSAGE I-D mandates advertising
>>of support in REGISTERs of a MESSAGE-capable UAC only *optionally*?
>
>Because the system will function without a proxy knowing this ahead of time. To date, we have not mandated the use of caller preferences (which is where this comes from) in other sip extensions, such as UPDATE, PRACK. I think presence mentions that you SHOULD do this, but not RFC 3265 (i.e.,  it doesnt apply to all events).
>
>Indeed, since not all methods are mandatory even in baseline SIP, and indeed, rfc3261 doesn't actually mandate anything but OPTIONS, we already have the case that a proxy doesn't know the methods of the UAs that register to it.
>
>>Making it optional gains little knowledge to a SIP server -- a server does not know anything it could reliably use in its processing logic.
>
>Sure it does. If it is there, it knows that the UA does or does not support MESSAGE. Its just that its not always there.
>
>>Think of a SIP server which acts as a proxy server and message store.
>>If it knew, that downstream UAS was not capable of processing MESSAGEs,
>>it could immediately store an incoming MESSAGE, send 202 upstream,
>>and relay the message to first UAC which registers and advertises support
>>for MESSAGE in REGISTER.
>>Because the server does not know it, it has to relay unnecessarily a message to an upstream UAS, which will deny it. 
>
>That doesn't seem so bad.
>
>If it bothers you, send an OPTIONS to the UA once it registers.
>
>
>
>>Then, whenever a REGISTER arrives, the server will try to dump message store to it,
>>(hoping a UAC capable of MESSAGE registered) resulting in quite
>>uneffective behaviour.
>
>Well, that is only one way to handle transmission of undelivered messages to a UA when it comes online. For many reasons, it is not the best way, I think. A better way is allow a UA to query to see if it has any undelivered messages when it registers, and then pull them as needed. This works better in wireless, too. Even if the UA does support MESSAGE, it may not want all of them the moment it connects.
>
>
>-Jonathan R.
>
>
>
>-- 
>Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
>Chief Scientist                         First Floor
>dynamicsoft                             East Hanover, NJ 07936
>jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
>http://www.jdrosen.net                  PH:  (973) 952-5000
>http://www.dynamicsoft.com
>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip


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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: Enhancements for Authenticated Identity Management in 
                          the Session Initiation Protocol (SIP)
	Author(s)	: J. Peterson
	Filename	: draft-peterson-sip-identity-01.txt
	Pages		: 29
	Date		: 05-Jul-02
	
The existing mechanisms for expressing identity in the Session
Initiation Protocol oftentimes do not permit an administrative domain
to verify securely the identity of the originator of a request.  This
document recommends practices and conventions for authenticating end
users, and proposes a way to distribute secure authenticated
identities within SIP messages.

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

	Title		: Security Mechanism Agreement for SIP Sessions
	Author(s)	: J. Arkko et al.
	Filename	: draft-ietf-sip-sec-agree-04.txt
	Pages		: 17
	Date		: 05-Jul-02
	
SIP has a number of security mechanisms. Some of them have been built
in to the SIP protocol, such as HTTP authentication or secure
attachments. These mechanisms have even alternative algorithms and
parameters. SIP does not currently provide any mechanism for
selecting which security mechanisms to use between two entities. In
particular, even if some mechanisms such as OPTIONS were used to make
this selection, the selection would be vulnerable against the
Bidding-Down attack. This document defines three header fields for
negotiating the security mechanisms within SIP between a SIP entity
and its next SIP hop. A SIP entity applying this mechanism must
always require some minimum security (i.e. integrity protection) from
all communicating parties in order to secure the negotiation, but the
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This draft is a work item of the Session Initiation Protocol Working Group of the IETF.

	Title		: Session Initiation Protocol (SIP) Caller Preferences 
                          and Callee Capabilities
	Author(s)	: H. Schulzrinne, J. Rosenberg
	Filename	: draft-ietf-sip-callerprefs-06.txt
	Pages		: 28
	Date		: 05-Jul-02
	
This document describes a set of extensions to the Session Initiation
Protocol (SIP) which allow a caller to express preferences about
request handling in servers. These preferences include the ability to
select which URIs a request gets routed to, and to specify certain
request handling directives in proxies and redirect servers. It does
so by defining three new request headers, Accept-Contact, Reject-
Contact and Request-Disposition, which specify the caller's
preferences. The extension also defines new parameters for the
Contact header that describe the characterstics of a UA.

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From daemon@optimus.ietf.org  Sun Jul  7 13:01:14 2002
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Date: Sun, 07 Jul 2002 12:33:05 -0400
Subject: [Sip] I-D ACTION:draft-ietf-sip-nat-02.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.

	Title		: An Extension to the Session Initiation Protocol (SIP) 
                          for Symmetric Response Routing
	Author(s)	: J. Rosenberg, J. Weinberger, H. Schulzrinne
	Filename	: draft-ietf-sip-nat-02.txt
	Pages		: 8
	Date		: 05-Jul-02
	
The Session Initiation Protocol (SIP) operates over UDP and TCP. When
used with UDP, responses to requests are returned to the source
address the request came from, but from the port written into the
topmost Via header of the request. This behavior is not desirable in
many cases, most notably, when the client is behind a NAT. This
extension defines a new parameter for the Via header, called rport,
that allows a client to request that the server send the response
back to the source IP address and port where the request came from.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-nat-02.txt

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_______________________________________________
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From daemon@optimus.ietf.org  Sun Jul  7 13:03:11 2002
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Date: Sun, 07 Jul 2002 12:33:08 -0400
Subject: [Sip] I-D ACTION:draft-ietf-sip-session-timer-09.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.

	Title		: Session Initiation Protocol Extension for Session 
                          Timer
	Author(s)	: S. Donovan, J. Rosenberg
	Filename	: draft-ietf-sip-session-timer-09.txt
	Pages		: 25
	Date		: 05-Jul-02
	
This document defines an extension to the Session Initiation Protocol
(SIP). This extension allows for a periodic refresh of SIP sessions
through a re-INVITE or UPDATE request. The refresh allows both user
agents and proxies to determine if the SIP session is still active.
The extension defines two new header fields, Session-Expires, which
conveys the lifetime of the session, and Min-SE, which conveys the
minimum allowed value for the session timer.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-session-timer-09.txt

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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Sun Jul  7 21:52:19 2002
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From: "Sriram Parameswar" <sriramp@nortelnetworks.com>
To: "'seanol@microsoft.com'" <seanol@microsoft.com>,
        "'sip@ietf.org'"
	 <sip@ietf.org>
Date: Sun, 7 Jul 2002 20:34:31 -0500 
MIME-Version: 1.0
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Subject: [Sip] draft-olson-sip-content-indirect-mech-00
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------_=_NextPart_001_01C2261F.9BB72D30
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Sean,

Well written draft!  In Section 3.6 'Specifying the type of the indirect
content' - it must be clearly noted that when including the content-type of
the indirect content in the payload - there must be Content headers
specifying that payload. Along with some textual description added to
section 3.6,  the example in that section will change to look something
like:

Content-Type: message/external-body; access-type="URL"; expiration="Mon, 24
June 2002 09:00:00 GMT"; URL="http://www.bogus.com/the-indirect-content" 
Content-Type: text/plain 
Content-Length: ...
<CRLF> 
Content-Type: application/sdp 
<CRLF> 
You may also want to add a small section on how this mechanism satisfies the
requirements laid down in the requirements document
(draft-ietf-sipping-content-indirect-00). Just as an after thought, the
additional Content-Type may also be message/sipfrag instead of text/plain.

Regards,

Sriram

__________________________________________
Sriram Parameswar              Phone: 972-685-8540
Interactive Multimedia Server (IMS) Fax: 972-684-3986
Nortel Networks, Richardson USA  Email: sriramp@nortelnetworks.com


------_=_NextPart_001_01C2261F.9BB72D30
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<HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2654.89">
<TITLE>draft-olson-sip-content-indirect-mech-00</TITLE>
</HEAD>
<BODY>

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

<P><FONT SIZE=3D2 FACE=3D"Arial">Well written draft!&nbsp; In</FONT> =
<FONT SIZE=3D2 FACE=3D"Arial">Section 3.6 'Specifying the type of the =
indirect content' - it must be clearly noted that when including the =
content-type of the indirect content in the payload - there must be =
Content headers specifying that payload. Along with some textual =
description added to section 3.6,&nbsp; the example in that section =
will change to look something like:</FONT></P>

<P><FONT FACE=3D"Times New Roman">Content-Type: message/external-body; =
access-type=3D&quot;URL&quot;; expiration=3D&quot;Mon, 24 June 2002 =
09:00:00 GMT&quot;; URL=3D&quot;<A =
HREF=3D"http://www.bogus.com/the-indirect-content" =
TARGET=3D"_blank">http://www.bogus.com/the-indirect-content</A>&quot; =
</FONT></P>

<P><FONT FACE=3D"Times New Roman">Content-Type: text/plain </FONT>
<BR><FONT FACE=3D"Times New Roman">Content-Length: ...</FONT>
<BR><FONT FACE=3D"Times New Roman">&lt;CRLF&gt; </FONT>
<BR><FONT FACE=3D"Times New Roman">Content-Type: application/sdp =
</FONT>
<BR><FONT FACE=3D"Times New Roman">&lt;CRLF&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">You may also want to add a small =
section on how this mechanism satisfies the requirements laid down in =
the requirements document (draft-ietf-sipping-content-indirect-00). =
Just as an after thought, the additional Content-Type may also be =
message/sipfrag instead of text/plain.</FONT></P>

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

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

<P><B><FONT SIZE=3D2 FACE=3D"Comic Sans =
MS">__________________________________________</FONT></B>
<BR><B><FONT SIZE=3D2 FACE=3D"Comic Sans MS">Sriram =
Parameswar&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D2 FACE=3D"Arial">Phone: =
972-685-8540</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Interactive Multimedia Server (IMS) =
Fax: 972-684-3986</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Nortel Networks, Richardson =
USA&nbsp;</FONT> <FONT SIZE=3D2 FACE=3D"Arial Narrow">Email: =
sriramp@nortelnetworks.com</FONT>
</P>

</BODY>
</HTML>
------_=_NextPart_001_01C2261F.9BB72D30--

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Mon Jul  8 01:34:34 2002
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From: Postmann Erwin <erwin.postmann@siemens.com>
To: "'sip@ietf.org'" <sip@ietf.org>
Date: Mon, 8 Jul 2002 07:12:41 +0200 
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Dear SIP experts,

I have a question regarding the support of multicast within SIP:

How is it possible to establish a connection towards a group of different
users in parallel? 

Is it possible to send a SIP INVITE/MESSAGE message to a multicast address,
in such a way that members of this multicast group can receive them in
parallel?



Best regards

Erwin Postmann

SIEMENS AG Austria
Program and System Engineering
Mobile Communication Systems 
Systems, Standardisation and Technology

Autokaderstrasse 29
A-1210 Vienna

Tel:	+43.5.1707.21398
Fax:	+43.5.1707.51924
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email:	erwin.postmann@siemens.com

Important Note: 
This e-mail may contain trade secrets or privileged, undisclosed or
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or distribution of it is strictly prohibited. Please inform us immediately
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From daemon@optimus.ietf.org  Mon Jul  8 04:24:54 2002
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Subject: Re: [Sip] Manyfolks questions
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Mark,

this is addressed in the second paragraph of page 10:

" Note that the rules above apply to the desired strength-tag "none" as
   well. This way, a user agent that supports quality of service but
   does not intend to use them, adds desired status lines with the
   strength-tag "none". Since this tag can be upgraded in the answer, as
   described in Section 5.2, the answerer can request quality of service
   reservation without a need of another offer/answer exchange.
"

This corresponds to your option number 2.

Gonzalo

Mark Watson wrote:
> 
> If the UAC knows of no preconditions for a session, but supports the
> manyfolks mechanism, what should it indicate in the INVITE in order to give
> the UAS the opportunity to indicate that it has (presumably local)
> preconditions ?
> 
> 1) Indicate 'Supported: preconditions' in the INVITE, but do not include
> a=cur: a=des: etc. in the SDP offer. Then the UAS can return the a=cur:
> a=des: in the SDP answer.
> 
> OR
> 
> 2) Indicate 'Supported: preconditions' in the INVITE and include the
> attributes but with desired status 'none'
> 
> If it's (1), we need description of the answerer adding the pre-condition
> attributes when they were not present in the offer.
> 
> If it's (2), then it should be mandatory to include the pre-condition
> attributes if you include 'Supported: preconditions'.
> 
> I couldn't find either in the specification, but perhaps I didn't look hard
> enough!
> 
> Regards,
> 
> Mark
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo

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From daemon@optimus.ietf.org  Mon Jul  8 07:00:30 2002
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Adam,

What you are describing is what I meant the draft to say. In addition, I
also agree with you that some clarification is needed, so I will add
some text about it.

Below you have the last paragraph of section 3. This paragraph deals
with a proxy/UAC sending the first INVITE:
"Regardless of whether the request is sent compressed or not, if a
   client would like to receive subsequent requests within the same
   dialog compressed, this client SHOULD add the parameter comp=sigcomp
   to the URI in the Contact header field if it is a user agent client.
   If the client is a proxy, it SHOULD add the parameter comp=sigcomp to
   its URI in the Record-Route header field."

the paragraph above is supposed to address what you reffer to as "a
proxy tailoring its Record-Route entry for B". I believe that if we do
the following addition the paragraph would be clearer.

OLD:
... requests within the same dialog compressed.

NEW:
... requests within the same dialog in the UAS->UAC direction
compressed.

I will also add that if a proxy/UAC sends a request compressed (e.g., a
proxy to a wireless terminals that had REGISTERED a comp=sigcomp SIP
URI), it SHOULD add comp=sigcomp to its RR entry.


The paragraph below is the second paragraph of section 4:
"  A proxy performing Record-Route inspects the Record-Route header
   field in the response and the Contact header field in the request
   that triggered this response (see example in Section 7). It looks for
   the URI of the next upstream (closer to the user agent client) hop in
   the route set. If this URI contains the parameter comp=sigcomp, the
   proxy SHOULD add comp=sigcomp to its entry in the Record-Route header
   field."

The paragraph above is supposed to address the rewriting of the
Record-Route entry in order to make it suitable for A's consumption. Let
me know if you think something should be clarified.


With the clarifications above, the example you gave will work just fine.

Some people have expressed concerns about the meaning of comp=sigcomp.
In the current draft it means *willingness* (clarification will be
added), whereas folks have proposed that it should mean only *support*.

If the parameter meant support, in scenarios like the one Adam showed:

> 
> +---+   +----+   +----+   +----+   +---+
> | A |...| P1 |---| P2 |---| P3 |...| B |
> +---+   +----+   +----+   +----+   +---+

P2 would know that P1 and P3 supported sigcomp, but it would have no
data available to make the decision about whether or not to send
compress messages towards both proxies. However, if the parameter means
willingness, it will be clear for P2 that it does not need to compress
any message. 

And talking into consideration that wireless terminals (3G or not) will
typically REGISTER a comp=sigcomp SIP URI, the last proxy (i.e., P3)
will typically know that it has to RR with comp=sigcomp in the first
INVITE when it consults the registrar.

Thanks for the feedback,

Gonzalo




Adam Roach wrote:
> 
> There has been a bit of discussion about how to handle
> Contact, Route, and Record-Route with the "comp=sigcomp"
> parameter.
> 
> There's a closely related issue that hasn't yet been touched
> on, though: when the called party issues a request, the
> "comp=sigcomp" headers end up being essentially in the wrong
> place.
> 
> I'll demonstrate by way of an example.
> 
> +---+   +----+   +----+   +----+   +---+
> | A |...| P1 |---| P2 |---| P3 |...| B |
> +---+   +----+   +----+   +----+   +---+
> 
> A and B are wireless terminals with very low bandwidth
> links (and, consequently, use SigComp). P1, P2, and P3 are
> all proxies on the network.
> 
> A sends an INVITE to B. Assuming that each proxy
> inserts its record-routes so that A gets the proper
> route-set, you get the following at B:
> 
> Record-Route: <sip:abcdef@p3>
> Record-Route: <sip:uvwxyz@p2>
> Record-Route: <sip:123456@p1;comp=sigcomp>
> Contact: <sip:usera@A;comp=sigcomp>
> 
> This is, of course, stored in B as the route set:
>   <sip:abcdef@p3>,<sip:uvwxyz@p2>,<sip:123456@p1;comp=sigcomp>,
>   <sip:usera@A;comp=sigcomp>
> 
> The response to the request delivers the same set of Record-Routes
> to A, but with a contact of "sip:userb@B;comp=sigcomp":
>   <sip:123456@p1;comp=sigcomp>,<sip:uvwxyz@p2>,<sip:abcdef@p3>,
>   <sip:userb@B;comp=sigcomp>
> 
> So, A's route is absolutely correct; it will compress the message
> sent to P1, and P3 will compress the message sent to B. Unfortunately,
> B's route is really screwy. If it obeys the "comp" parameters, B
> *won't* compress any requests in this dialog, so you'll be sending
> uncompressed messages over the radio link. And, in a bitter irony,
> P2 will waste cycles compressing the messages sent to P1 -- which
> turns out to be completely unnecessary. In fact, it exacerbates the
> fact that the B -> P3 request was sent uncompressed.
> 
> If we have the proxies tailor the route set for B, the same problem
> happens, in the opposite direction.
> 
> The solution lies in step 8, detailed in section 16.7 of
> 2543bis-09.
> 
> In this case, the proxies would tailor the inserted "Record-Routes"
> in the request for B's consumption:
> 
> Record-Route: <sip:abcdef@p3;comp=sigcomp>
> Record-Route: <sip:uvwxyz@p2>
> Record-Route: <sip:123456@p1>
> Contact: <sip:usera@A;comp=sigcomp>
> 
> Then, as the INVITE response passes back through P3 and P1, they
> would each modify the Record-Route headers so that they are
> appropriate for A's consumption:
> 
> Record-Route: <sip:abcdef@p3>
> Record-Route: <sip:uvwxyz@p2>
> Record-Route: <sip:123456@p1;comp=sigcomp>
> Contact: <sip:userb@B;comp=sigcomp>
> 
> Even though it's described in SIP, this usage should probably be
> explicitly mentioned in in the SIP SigComp draft; otherwise, we
> run the risk of having nodes do strange things.
> 
> /a

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CC: "'Alexeitsev, D'" <D.Alexeitsev@telekom.de>, sip@ietf.org
Subject: Re: [Sip] Local vs Remote ringback on 180
References: <000701c22370$a9001760$6600a8c0@bculpepperpc>
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Bert Culpepper wrote:
> 
> Hi Denis,
> 
> I don't know that the use of TCP is a substitute for 100rel.  PRACK
> provides knowledge that the receiver's UA "processed" the provisional
> response.  TCP only guarantees the message was received at the host.  I
> may be off-base here, if so others can correct me.

PRACK is end-to-end. TCP/UDP/SCTP is hop-by-hop. That is, if you send a
response using TCP you know that it will reach the next proxy, but you
know nothing about the remote UA.

About special casing the scenario where all the Via headers show that
TCP is used, it would be adding extra complexity for a little gain
(KISS). In addition, think of the interactions with remote Contact
updates from TCP to UDP using re-INVITEs.

Regards,

Gonzalo

> 
> Regards,
> Bert
> 
> Alexeitsev, D wrote:
> >
> > Hello,
> >
> > comment inline
> >
> > > [DA] Is the use of the TCP/SCTP is good enough to say that
> > > e2e provisional responses with answer are reliable?
> >
> > I don't think so for "reliable provisional responses".
> > Retransmissions
> > are not needed when using TCP but a PRACK from the UAC is still needed
> > for "reliable provisional responses".
> >
> > [DA] I mean, that the 100rel is only needed for the UDP. The
> > reliable transport protocols do not need a help from session
> > control protocols.
> >
> > Greetings,
> > Denis Alexeitsev
> >
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo

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



From daemon@optimus.ietf.org  Mon Jul  8 08:24:48 2002
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Subject: [Sip] Authentication for re-INVITE.
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Is a SIP proxy supposed to Authenticate a re-INVITE also? If no, then if
some proxy in Authenticating and it asks for Auth Info from the UA, what
is the UA supposed to do? Is it mandatory that it should send the Auth
Info?

Please suggest!

Thanks & Regards
s

Ph.: 091-124-6383474 (e 250)
__________________________________________
I had a good grip on life, but then the handle broke.

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From daemon@optimus.ietf.org  Mon Jul  8 09:40:14 2002
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From: "Ranjit Avasarala" <ranjit.avasarala@bplmail.com>
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Subject: Re: [Sip] Authentication for re-INVITE.
Date: Mon, 8 Jul 2002 18:30:00 +0530
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I don't think proxy needs to authenticate re-invite, as re-invite is used
for changing session parameters of an existing call

Ranjit
----- Original Message -----
From: ShavinderPal Singh <ShavinderPal.Singh@esamsungumit.com>
To: <sip@ietf.org>; Jonathan Rosemberg <jdrosen@dynamicsoft.com>
Sent: Monday, July 08, 2002 5:21 PM
Subject: [Sip] Authentication for re-INVITE.


>
> Is a SIP proxy supposed to Authenticate a re-INVITE also? If no, then if
> some proxy in Authenticating and it asks for Auth Info from the UA, what
> is the UA supposed to do? Is it mandatory that it should send the Auth
> Info?
>
> Please suggest!
>
> Thanks & Regards
> s
>
> Ph.: 091-124-6383474 (e 250)
> __________________________________________
> I had a good grip on life, but then the handle broke.
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Mon Jul  8 09:48:19 2002
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From: "Ranjit Avasarala" <ranjit.avasarala@bplmail.com>
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Subject: [Sip] sip phone query
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Hi
     Like when i want to register any event in server, I use SUBSCRIBE
method. How is this realized in SIP Phone? How do I specify event name, etc,
using a SIP Phone? Will there be any key or should I dial any digit...

Thanks
Ranjit


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From daemon@optimus.ietf.org  Mon Jul  8 12:11:31 2002
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From: Adam Roach <adam@dynamicsoft.com>
To: "'ShavinderPal Singh'" <ShavinderPal.Singh@esamsungumit.com>
Cc: "'sip@ietf.org'" <sip@ietf.org>
Subject: RE: [Sip] Authentication for re-INVITE.
Date: Mon, 8 Jul 2002 11:46:27 -0400 
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> -----Original Message-----
> From: ShavinderPal Singh [mailto:ShavinderPal.Singh@esamsungumit.com]
>
> Is a SIP proxy supposed to Authenticate a re-INVITE also?

This is really a question for the sip-implementors mailing list.
See the bottom of this message (and every other message on this
mailing list) for further details.

/a

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From daemon@optimus.ietf.org  Mon Jul  8 16:10:04 2002
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Date: Mon, 08 Jul 2002 14:22:33 -0500
From: Ben Campbell <bcampbell@dynamicsoft.com>
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Subject: [Sip] Re: MESSAGE advertising in REGISTERs
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I do not understand why we would expect support of MESSAGE to give a 
client a higher requirement for registration than for clients that do 
not support MESSAGE.

Jiri Kuthan wrote:
> Though advertising MESSAGE capability in REGISTERs does
> not hurt in any scenario, not advertising hurts in some.
> That's why I would much prefer having it as a MUST.
> 
> -Jiri
> 
> At 06:51 PM 7/3/2002, Ben Campbell wrote:
> 
>>Certainly it will be needed to make many server related scenarios work. It is optional because MESSAGE can also be used in a pure end-to-end mode without assuming that such servers exist. For example, a MESSAGE UAC may discover the contact for a MESSAGE UAS from presence information (i.e. a contact in a presence doc), then send the MESSAGE request directly with no intervening servers. In this architecture, there is not necessarily a need to advertise support at a registrar.
>>
>>Jiri Kuthan wrote:
>>
>>>Can someone tell me why the MESSAGE I-D mandates advertising
>>>of support in REGISTERs of a MESSAGE-capable UAC only *optionally*?
>>>Making it optional gains little knowledge to a SIP server -- a server does not know anything it could reliably use in its processing logic.
>>>Think of a SIP server which acts as a proxy server and message store.
>>>If it knew, that downstream UAS was not capable of processing MESSAGEs,
>>>it could immediately store an incoming MESSAGE, send 202 upstream,
>>>and relay the message to first UAC which registers and advertises support
>>>for MESSAGE in REGISTER.
>>>Because the server does not know it, it has to relay unnecessarily a message to an upstream UAS, which will deny it. Then, whenever a REGISTER arrives, the server will try to dump message store to it,
>>>(hoping a UAC capable of MESSAGE registered) resulting in quite
>>>uneffective behaviour.
>>>-Jiri
>>




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From daemon@optimus.ietf.org  Tue Jul  9 04:07:35 2002
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To: Salva Rey <salreyca@hotmail.com>
CC: sip@ietf.org
Subject: Re: [Sip] Forking proxy
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Since the proxy will CANCEL all the branches but the one that sent the
final response, you do not really care about what happens with that
provisional response that got into the race condition.

Besides, the idea of reliable provisional responses is that the proxy is
not aware of whether or not a particular response was sent reliably. The
reliability mechanism is end-to-end.

Gonzalo

Salva Rey wrote:
> 
> Hi all,
> 
> a doubt regarding forking proxies,
> 
> why would a proxy do if forked an INVITE request, it got back some reliable
> provisional responses that established some early dialogs at the UAC, and
> then one of the UAS's sends a final response 200 oK. If the other early
> dialogs try to send further reliable provisional responses this wont get
> through the proxy since a final response was already sent on the response
> context. Should the proxy behaviour not only forward immediately 200ok
> responses to INVITE, but also reliable responses ?
> 
> best regards
> Salva
> 
> _________________________________________________________________
> MSN Photos is the easiest way to share and print your photos:
> http://photos.msn.com/support/worldwide.aspx
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo

_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Jul  9 04:31:05 2002
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From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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Subject: [Sip] Re: [Sipping] draft-mahy-sipping-connect-reuse-reqs-00
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Sean,

Sean Olson wrote:
> 
> This is a very interesting problem. Should the requirements
> include the ability to deal with intermediaries such as
> NAT/firewalls that may not strictly be SIP entities? Also, I
> wonder if the proxy-proxy connection problem is as critical
> as the UAC-proxy connection reuse problem. Might SCTP be a
> better solution for proxy-proxy connection management?

Both SCTP and TCP have the problem described in this draft, IMO.

However, this problem would be solvable providing the proper port number
in the Record-Route entry. Assuming that SIP implementations check if
they have an existing connection with a transport address (destination
IP address and port number) before opening another one, everything would
work fine.

I see that the mechanism described in the draft has the advantage of
keeping the sender from worring about which ephemeral port is used to
send the request. Is there any other advantage if compared with
Record-Route entries with port numbers? If so, it should be discussed in
the draft.

In addition, I would add to the section about the problem statement that
if two SIP entities establish two TCP/SCTP connections rather than one,
the congestion avoidance data is not shared by the connections. This may
trigger a too aggressive behavior or a too conservative behavior ehrn
compared to the one achieved by using only one connection.

Gonzalo

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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From daemon@optimus.ietf.org  Tue Jul  9 04:59:03 2002
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Date: Tue, 09 Jul 2002 11:30:22 +0300
From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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        sipping@ietf.org
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Subject: [Sip] Re: [Sipping] draft-mahy-sipping-connect-reuse-reqs-00
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Before I had missed that this draft is supposed to fill the role of the
old Translate header in the old sip-nat draft. I believe that this
should be mentioned, since it is an important advantage over the
Record-Route mechanism.

Gonzalo

Gonzalo Camarillo wrote:
> 
> Sean,
> 
> Sean Olson wrote:
> >
> > This is a very interesting problem. Should the requirements
> > include the ability to deal with intermediaries such as
> > NAT/firewalls that may not strictly be SIP entities? Also, I
> > wonder if the proxy-proxy connection problem is as critical
> > as the UAC-proxy connection reuse problem. Might SCTP be a
> > better solution for proxy-proxy connection management?
> 
> Both SCTP and TCP have the problem described in this draft, IMO.
> 
> However, this problem would be solvable providing the proper port number
> in the Record-Route entry. Assuming that SIP implementations check if
> they have an existing connection with a transport address (destination
> IP address and port number) before opening another one, everything would
> work fine.
> 
> I see that the mechanism described in the draft has the advantage of
> keeping the sender from worring about which ephemeral port is used to
> send the request. Is there any other advantage if compared with
> Record-Route entries with port numbers? If so, it should be discussed in
> the draft.
> 
> In addition, I would add to the section about the problem statement that
> if two SIP entities establish two TCP/SCTP connections rather than one,
> the congestion avoidance data is not shared by the connections. This may
> trigger a too aggressive behavior or a too conservative behavior ehrn
> compared to the one achieved by using only one connection.
> 
> Gonzalo
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Jul  9 10:01:32 2002
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From: Christer Holmberg <christer.holmberg@lmf.ericsson.se>
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CC: Sean Olson <seanol@windows.microsoft.com>, rohan@cisco.com, sip@ietf.org,
        sipping@ietf.org
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Hi,

RFC3261/bis-9 mainly talks about using a single connection (or, one in each
direction) for a single transaction, and the connect-reuse draft talks about
being able to use a single connection for messages in both direction, for a
single session. Sometimes, however, when we talk about "reuse of connection"
we mean when a single connection (or, one in each direction) is used for
multiple sessions. I think it would be a good idea to mention that that kind
of "reuse" is out of the scope of the draft. Otherwise the draft subject
should more clear indicate what kind of reuse it refers to...

Of course, once a single connection is established, according to the
connect-reuse draft, it MAY be used for multiple sessions, but that's not the
scope of the draft, so...

Regards,

Christer Holmberg
Ericsson Finland



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This list is for NEW development of the core SIP Protocol
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From daemon@optimus.ietf.org  Tue Jul  9 10:01:43 2002
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To: Ranjit Avasarala <ranjit.avasarala@bplmail.com>
CC: ShavinderPal Singh <ShavinderPal.Singh@esamsungumit.com>, sip@ietf.org,
        Jonathan Rosemberg <jdrosen@dynamicsoft.com>
Subject: Re: [Sip] Authentication for re-INVITE.
References: <89CFEE6120A0224EA07781892EC0B48B899749@esuims01.esamsungumit.com> <016a01c2267f$5e29e060$5f0d0a0a@ranjit>
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Ranjit Avasarala wrote:
> 
> I don't think proxy needs to authenticate re-invite, as re-invite is used
> for changing session parameters of an existing call

For this very reason you should reauthenticate...so that someone doesnt
spoof a forged INVITE with bad parameters within it back to the client.

> 
> Ranjit
> ----- Original Message -----
> From: ShavinderPal Singh <ShavinderPal.Singh@esamsungumit.com>
> To: <sip@ietf.org>; Jonathan Rosemberg <jdrosen@dynamicsoft.com>
> Sent: Monday, July 08, 2002 5:21 PM
> Subject: [Sip] Authentication for re-INVITE.
> 
> >
> > Is a SIP proxy supposed to Authenticate a re-INVITE also? If no, then if
> > some proxy in Authenticating and it asks for Auth Info from the UA, what
> > is the UA supposed to do? Is it mandatory that it should send the Auth
> > Info?
> >
> > Please suggest!
> >
> > Thanks & Regards
> > s
> >
> > Ph.: 091-124-6383474 (e 250)
> > __________________________________________
> > I had a good grip on life, but then the handle broke.
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Jul  9 10:51:09 2002
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From: KANUMURI Sreeram <SKanumur@covansys.com>
To: "'chris@infravast.com'" <chris@infravast.com>,
        Ranjit Avasarala
	 <ranjit.avasarala@bplmail.com>
Cc: ShavinderPal Singh <ShavinderPal.Singh@esamsungumit.com>, sip@ietf.org
Subject: RE: [Sip] Authentication for re-INVITE.
Date: Tue, 9 Jul 2002 10:19:37 -0400 
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> 
> I don't think proxy needs to authenticate re-invite, as re-invite is used
> for changing session parameters of an existing call

>>For this very reason you should reauthenticate...so that someone doesnt
>>spoof a forged INVITE with bad parameters within it back to the client.

TRUE but can you tell me how can "someone  spoof a forged INVITE" ??
like the call is established between the end users and it is olny session
parameters change.
As there is Already the authentication done.
olny the UA in the call can change teh parameters with a new invite.....??

so can u make me  clearer??

Sreeram

> 
> Ranjit
> ----- Original Message -----
> From: ShavinderPal Singh <ShavinderPal.Singh@esamsungumit.com>
> To: <sip@ietf.org>; Jonathan Rosemberg <jdrosen@dynamicsoft.com>
> Sent: Monday, July 08, 2002 5:21 PM
> Subject: [Sip] Authentication for re-INVITE.
> 
> >
> > Is a SIP proxy supposed to Authenticate a re-INVITE also? If no, then if
> > some proxy in Authenticating and it asks for Auth Info from the UA, what
> > is the UA supposed to do? Is it mandatory that it should send the Auth
> > Info?
> >
> > Please suggest!
> >
> > Thanks & Regards
> > s
> >
> > Ph.: 091-124-6383474 (e 250)
> > __________________________________________
> > I had a good grip on life, but then the handle broke.
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip

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From daemon@optimus.ietf.org  Tue Jul  9 12:55:26 2002
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From: Christer Holmberg <christer.holmberg@lmf.ericsson.se>
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To: Sean Olson <seanol@windows.microsoft.com>
CC: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>, rohan@cisco.com,
        sip@ietf.org, sipping@ietf.org
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Subject: [Sip] Re: [Sipping] draft-mahy-sipping-connect-reuse-reqs-00
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Hi Sean,

> Is this really out of scope for this draft? Naturally
> the more re-use you make of the connection, the more
> benefit you get from re-use ;-)

Of course, but in that case it should be mentioned in the draft :)

The only thing the draft currently talks about the problem how to make sure
a single connection is used for both directions, per session - which is a
separate issue from reusing connections for multiple sessions.

And, since I don't think any draft currently talks about the reuse of
connections for multiple sessions (bis talks about keeping the connection
open to make sure the transaction uses the same transaction), maybe it
could be added to the connect-reuse draft?

Just some thoughts...

Regards,

Christer Holmberg
Ericsson Finland



>
>
> /sean
>
> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@lmf.ericsson.se]
> Sent: Tuesday, July 09, 2002 6:30 AM
> To: Gonzalo Camarillo
> Cc: Sean Olson; rohan@cisco.com; sip@ietf.org; sipping@ietf.org
> Subject: Re: [Sipping] draft-mahy-sipping-connect-reuse-reqs-00
>
> Hi,
>
> RFC3261/bis-9 mainly talks about using a single connection (or, one in
> each
> direction) for a single transaction, and the connect-reuse draft talks
> about being able to use a single connection for messages in both
> direction, for a single session. Sometimes, however, when we talk about
> "reuse of connection" we mean when a single connection (or, one in each
> direction) is used for multiple sessions. I think it would be a good
> idea to mention that that kind of "reuse" is out of the scope of the
> draft. Otherwise the draft subject should more clear indicate what kind
> of reuse it refers to...
>
> Of course, once a single connection is established, according to the
> connect-reuse draft, it MAY be used for multiple sessions, but that's
> not the scope of the draft, so...
>
> Regards,
>
> Christer Holmberg
> Ericsson Finland


_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Jul  9 13:07:37 2002
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From: Christer Holmberg <christer.holmberg@lmf.ericsson.se>
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Subject: [Sip] Re: [Sipping] draft-mahy-sipping-connect-reuse-reqs-00
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Another comment,

I also think it is important to remember that there MAY be scenarios,
especially if a connection is used for multiple sessions, where one wants to
use separate connections per each direction - even if the client indicates he
wants to use the same connection for the response. Just to think of before the
MUSTs are added...

Regards,

Christer Holmberg
Ericsson Finland


Christer Holmberg wrote:

> Hi Sean,
>
> > Is this really out of scope for this draft? Naturally
> > the more re-use you make of the connection, the more
> > benefit you get from re-use ;-)
>
> Of course, but in that case it should be mentioned in the draft :)
>
> The only thing the draft currently talks about the problem how to make sure
> a single connection is used for both directions, per session - which is a
> separate issue from reusing connections for multiple sessions.
>
> And, since I don't think any draft currently talks about the reuse of
> connections for multiple sessions (bis talks about keeping the connection
> open to make sure the transaction uses the same transaction), maybe it
> could be added to the connect-reuse draft?
>
> Just some thoughts...
>
> Regards,
>
> Christer Holmberg
> Ericsson Finland
>
> >
> >
> > /sean
> >
> > -----Original Message-----
> > From: Christer Holmberg [mailto:christer.holmberg@lmf.ericsson.se]
> > Sent: Tuesday, July 09, 2002 6:30 AM
> > To: Gonzalo Camarillo
> > Cc: Sean Olson; rohan@cisco.com; sip@ietf.org; sipping@ietf.org
> > Subject: Re: [Sipping] draft-mahy-sipping-connect-reuse-reqs-00
> >
> > Hi,
> >
> > RFC3261/bis-9 mainly talks about using a single connection (or, one in
> > each
> > direction) for a single transaction, and the connect-reuse draft talks
> > about being able to use a single connection for messages in both
> > direction, for a single session. Sometimes, however, when we talk about
> > "reuse of connection" we mean when a single connection (or, one in each
> > direction) is used for multiple sessions. I think it would be a good
> > idea to mention that that kind of "reuse" is out of the scope of the
> > draft. Otherwise the draft subject should more clear indicate what kind
> > of reuse it refers to...
> >
> > Of course, once a single connection is established, according to the
> > connect-reuse draft, it MAY be used for multiple sessions, but that's
> > not the scope of the draft, so...
> >
> > Regards,
> >
> > Christer Holmberg
> > Ericsson Finland
>
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP


_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Jul  9 13:40:40 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
To: <sip@ietf.org>
Cc: <jo@ipdialog.com>, <brian.rosen@marconi.com>, <rohan@cisco.com>
Date: Tue, 9 Jul 2002 12:19:52 -0500
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Subject: [Sip] Announcing WGLC for draft-ietf-sip-session-timer
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I'd like to request working group last call on the session timer draft.

This should be pretty straightforward -- the -08 version was WGLCed,
sent up for IETF last call, then held up waiting on RFC3261. The current
-09 version has been updated for RFC3261 compliance.

See: 

http://www.ietf.org/internet-drafts/draft-ietf-sip-session-timer-09.txt
Or
http://www.softarmor.com/sipwg/drafts/draft-ietf-sip-session-timer-09.tx
t


I think we should be able to close this up by July 22.

Thanks,

--
Dean


_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Jul  9 13:54:21 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
To: <sip@ietf.org>
Cc: <rohan@cisco.com>, <jo@ipdialog.com>, <brian.rosen@marconi.com>,
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Subject: [Sip] Announce WGLC on draft-ietf-sip-guidelines
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I'd like to request that the SIP working group review the SIP Guidelines
draft.

See: 

http://www.ietf.org/internet-drafts/draft-ietf-sip-guidelines-05.txt
Or
http://www.softarmor.com/sipwg/drafts/draft-ietf-sip-guidelines-05.txt


We should be able to close this the week of July 22.

Thanks

--
Dean


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



From daemon@optimus.ietf.org  Tue Jul  9 13:59:36 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
To: <sip@ietf.org>
Cc: <rohan@cisco.com>, <jo@ipdialog.coml>, <brian.rosen@marconi.com>,
        <gonzalo.camarillo@ericsson.com>, <jdrosen@dynamicsoft.com>
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Subject: [Sip] Anounce WGLC for draft-ietf-sip-sctp
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Please review the SIP over SCTP draft.

See:

http://www.ietf.org/internet-drafts/draft-ietf-sip-sctp-03.txt
Or
http://www.softarmor.com/sipwg/drafts/draft-ietf-sip-sctp-03.txt

As usual, please copy comments to the authors and/or the list as
appropriate.

I'd like to wrap this up by the week of July 22.

Thanks,

--
Dean


_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Jul  9 14:04:09 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
To: <sip@ietf.org>
Cc: <rohan@cisco.com>, <jo@ipdialog.com>, <brian.rosen@marconi.com>,
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Subject: [Sip] Anounce WGLC for draft-ietf-sip-callerpref
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I'd like to request the working group do a last call on the caller
preferences draft. 


See:

http://www.ietf.org/internet-drafts/draft-ietf-sip-callerprefs-06.txt
Or
http://www.softarmor.com/sipwg/drafts/draft-ietf-sip-callerprefs-06.txt

If possible, I'd like to close this by July 22. Please copy the author
and/or list with your comments as appropriate.

--
Dean


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This list is for NEW development of the core SIP Protocol
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From daemon@optimus.ietf.org  Tue Jul  9 14:08:52 2002
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From: "Sean Olson" <seanol@windows.microsoft.com>
To: "Christer Holmberg" <christer.holmberg@lmf.ericsson.se>,
        "Gonzalo Camarillo" <Gonzalo.Camarillo@lmf.ericsson.se>,
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Good point. Is this is a request or a demand by the client to re-use
the connection? I think we need the possibility for the client to
*demand* that the connection be re-used. There may also be situations
where re-use is just an optimization.

/sean

-----Original Message-----
From: Christer Holmberg [mailto:christer.holmberg@lmf.ericsson.se] 
Sent: Tuesday, July 09, 2002 9:41 AM
To: Sean Olson; Gonzalo Camarillo; rohan@cisco.com; sip@ietf.org;
sipping@ietf.org
Subject: Re: [Sipping] draft-mahy-sipping-connect-reuse-reqs-00



Another comment,

I also think it is important to remember that there MAY be scenarios,
especially if a connection is used for multiple sessions, where one
wants to use separate connections per each direction - even if the
client indicates he wants to use the same connection for the response.
Just to think of before the MUSTs are added...

Regards,

Christer Holmberg
Ericsson Finland


Christer Holmberg wrote:

> Hi Sean,
>
> > Is this really out of scope for this draft? Naturally
> > the more re-use you make of the connection, the more benefit you get

> > from re-use ;-)
>
> Of course, but in that case it should be mentioned in the draft :)
>
> The only thing the draft currently talks about the problem how to make

> sure a single connection is used for both directions, per session - 
> which is a separate issue from reusing connections for multiple 
> sessions.
>
> And, since I don't think any draft currently talks about the reuse of 
> connections for multiple sessions (bis talks about keeping the 
> connection open to make sure the transaction uses the same 
> transaction), maybe it could be added to the connect-reuse draft?
>
> Just some thoughts...
>
> Regards,
>
> Christer Holmberg
> Ericsson Finland
>
> >
> >
> > /sean
> >
> > -----Original Message-----
> > From: Christer Holmberg [mailto:christer.holmberg@lmf.ericsson.se]
> > Sent: Tuesday, July 09, 2002 6:30 AM
> > To: Gonzalo Camarillo
> > Cc: Sean Olson; rohan@cisco.com; sip@ietf.org; sipping@ietf.org
> > Subject: Re: [Sipping] draft-mahy-sipping-connect-reuse-reqs-00
> >
> > Hi,
> >
> > RFC3261/bis-9 mainly talks about using a single connection (or, one 
> > in each
> > direction) for a single transaction, and the connect-reuse draft 
> > talks about being able to use a single connection for messages in 
> > both direction, for a single session. Sometimes, however, when we 
> > talk about "reuse of connection" we mean when a single connection 
> > (or, one in each
> > direction) is used for multiple sessions. I think it would be a good
> > idea to mention that that kind of "reuse" is out of the scope of the
> > draft. Otherwise the draft subject should more clear indicate what
kind
> > of reuse it refers to...
> >
> > Of course, once a single connection is established, according to the

> > connect-reuse draft, it MAY be used for multiple sessions, but 
> > that's not the scope of the draft, so...
> >
> > Regards,
> >
> > Christer Holmberg
> > Ericsson Finland
>
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP Use 
> sip-implementors@cs.columbia.edu for questions on current sip Use 
> sip@ietf.org for new developments of core SIP



_______________________________________________
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From daemon@optimus.ietf.org  Tue Jul  9 14:11:28 2002
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Is this really out of scope for this draft? Naturally
the more re-use you make of the connection, the more
benefit you get from re-use ;-) 

/sean

-----Original Message-----
From: Christer Holmberg [mailto:christer.holmberg@lmf.ericsson.se] 
Sent: Tuesday, July 09, 2002 6:30 AM
To: Gonzalo Camarillo
Cc: Sean Olson; rohan@cisco.com; sip@ietf.org; sipping@ietf.org
Subject: Re: [Sipping] draft-mahy-sipping-connect-reuse-reqs-00



Hi,

RFC3261/bis-9 mainly talks about using a single connection (or, one in
each
direction) for a single transaction, and the connect-reuse draft talks
about being able to use a single connection for messages in both
direction, for a single session. Sometimes, however, when we talk about
"reuse of connection" we mean when a single connection (or, one in each
direction) is used for multiple sessions. I think it would be a good
idea to mention that that kind of "reuse" is out of the scope of the
draft. Otherwise the draft subject should more clear indicate what kind
of reuse it refers to...

Of course, once a single connection is established, according to the
connect-reuse draft, it MAY be used for multiple sessions, but that's
not the scope of the draft, so...

Regards,

Christer Holmberg
Ericsson Finland




_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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From daemon@optimus.ietf.org  Tue Jul  9 14:53:15 2002
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From: "Sriram Parameswar" <sriramp@nortelnetworks.com>
To: "'Sean Olson'" <seanol@windows.microsoft.com>, sip@ietf.org
Date: Tue, 9 Jul 2002 13:21:31 -0500 
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Subject: [Sip] RE: draft-olson-sip-content-indirect-mech-00
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This message is in MIME format. Since your mail reader does not understand
this format, some or all of this message may not be legible.

------_=_NextPart_001_01C22775.73498490
Content-Type: text/plain;
	charset="iso-8859-1"

Thanks Sean - I would definitely like to hear from the MIME experts - but
here is an example from RFC 2017 section 3 which is more along what I
suggested. I think we need to take a closer look because the current draft
does borrow heavily from RFC2017:

<quote RFC2017>
    Content-type: message/external-body; access-type=URL;
                  URL="http://www.foo.com/file"

    Content-type: text/html
    Content-Transfer-Encoding: binary

    THIS IS NOT REALLY THE BODY!

</quote RFC2017>

Regards,

Sriram

__________________________________________
Sriram Parameswar              Phone: 972-685-8540
Interactive Multimedia Server (IMS) Fax: 972-684-3986
Nortel Networks, Richardson USA  Email: sriramp@nortelnetworks.com


-----Original Message-----
From: Sean Olson [mailto:seanol@windows.microsoft.com]
Sent: Tuesday, July 09, 2002 12:38 PM
To: Parameswar, Sriram [NGC:B615:EXCH]; sip@ietf.org
Subject: RE: draft-olson-sip-content-indirect-mech-00


Thanks for the comments! 

I will try to dissect my original example to explain what I was
shooting for. I've added line numbers, plus some surrounding context
to hopefully make things a little clearer.

1   INVITE sip:bob@acme.com SIP/2.0
2   Via: SIP/2.0/TCP 10.0.0.2;branch=z9hG4bK83749
3   From: <sip:alice@acme.com>;tag=abcd
4   To: <sip:bob@acme.com>
5   Call-ID: 1@10.0.0.2
6   CSeq: 2343 INVITE
7   Content-Type: message/external-body; access-type="URL";
8                 expiration="Mon, 24 June 2002 09:00:00 GMT";
9                 URL="http://www.bogus.com/the-indirect-content"
10  Content-Length: ...
11  <CRLF>
12  Content-Type: application/sdp
13  <CRLF>


Line 12 is the start of the payload for this INVITE.
Lines 7-9 define the content type of this payload (indirect content
          available at the given URL)

Line 12 gives the content type of the indirect content. It is actually
        an entity header inside the payload (an encapsulated entity)

Line 13 marks the end of the entity headers for this encapsulated entity
        and also marks the end of the message

This is consistent with the example given in section 5.2.3 of RFC2046.
I'm using MIME in the broadest sense defined in RFC2045/2046 to allow
nesting or encapsulation of entities. This looks a bit strange compared
to other SIP messages, but I believe it is perfectly valid MIME. I
don't think we need to use message/sipfrag or even text/plain to
accomplish
this -- the capability to express nested entities and entity headers is
part of MIME.
I would be curious to hear from other MIME experts on this issue to see
if I'm
abusing something here.

/sean

-----Original Message-----
From: Sriram Parameswar [mailto:sriramp@nortelnetworks.com] 
Sent: Sunday, July 07, 2002 6:35 PM
To: Sean Olson; 'sip@ietf.org'
Subject: draft-olson-sip-content-indirect-mech-00


Sean, 
Well written draft!  In Section 3.6 'Specifying the type of the indirect
content' - 
it must be clearly noted that when including the content-type of the
indirect content in the payload
- there must be Content headers specifying that payload. Along with some
textual description added to section 3.6,
the example in that section will change to look something like:

Content-Type: message/external-body; access-type="URL"; expiration="Mon,
24 June 2002 09:00:00 GMT";
URL="http://www.bogus.com/the-indirect-content" 
Content-Type: text/plain 
Content-Length: ... 
<CRLF> 
Content-Type: application/sdp 
<CRLF> 

You may also want to add a small section on how this mechanism satisfies
the requirements laid down in the requirements document
(draft-ietf-sipping-content-indirect-00). Just as an after thought, the
additional Content-Type may also be message/sipfrag instead of
text/plain.

Regards, 
Sriram 
__________________________________________ 
Sriram Parameswar              Phone: 972-685-8540 
Interactive Multimedia Server (IMS) Fax: 972-684-3986 
Nortel Networks, Richardson USA  Email: sriramp@nortelnetworks.com 

------_=_NextPart_001_01C22775.73498490
Content-Type: text/html;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2654.89">
<TITLE>RE: draft-olson-sip-content-indirect-mech-00</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>Thanks Sean - I would definitely like to hear from =
the MIME experts - but here is an example from RFC 2017 section 3 which =
is more along what I suggested. I think we need to take a closer look =
because the current draft does borrow heavily from RFC2017:</FONT></P>

<P><FONT SIZE=3D2>&lt;quote RFC2017&gt;</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp; Content-type: =
message/external-body; access-type=3DURL;</FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; URL=3D&quot;<A =
HREF=3D"http://www.foo.com/file" =
TARGET=3D"_blank">http://www.foo.com/file</A>&quot;</FONT>
</P>

<P><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp; Content-type: text/html</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp; Content-Transfer-Encoding: =
binary</FONT>
</P>

<P><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp; THIS IS NOT REALLY THE =
BODY!</FONT>
</P>

<P><FONT SIZE=3D2>&lt;/quote RFC2017&gt;</FONT>
</P>

<P><FONT SIZE=3D2>Regards,</FONT>
</P>

<P><FONT SIZE=3D2>Sriram</FONT>
</P>

<P><FONT SIZE=3D2>__________________________________________</FONT>
<BR><FONT SIZE=3D2>Sriram =
Parameswar&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp; Phone: 972-685-8540</FONT>
<BR><FONT SIZE=3D2>Interactive Multimedia Server (IMS) Fax: =
972-684-3986</FONT>
<BR><FONT SIZE=3D2>Nortel Networks, Richardson USA&nbsp; Email: =
sriramp@nortelnetworks.com</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Sean Olson [<A =
HREF=3D"mailto:seanol@windows.microsoft.com">mailto:seanol@windows.micro=
soft.com</A>]</FONT>
<BR><FONT SIZE=3D2>Sent: Tuesday, July 09, 2002 12:38 PM</FONT>
<BR><FONT SIZE=3D2>To: Parameswar, Sriram [NGC:B615:EXCH]; =
sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>Subject: RE: =
draft-olson-sip-content-indirect-mech-00</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Thanks for the comments! </FONT>
</P>

<P><FONT SIZE=3D2>I will try to dissect my original example to explain =
what I was</FONT>
<BR><FONT SIZE=3D2>shooting for. I've added line numbers, plus some =
surrounding context</FONT>
<BR><FONT SIZE=3D2>to hopefully make things a little clearer.</FONT>
</P>

<P><FONT SIZE=3D2>1&nbsp;&nbsp; INVITE sip:bob@acme.com SIP/2.0</FONT>
<BR><FONT SIZE=3D2>2&nbsp;&nbsp; Via: SIP/2.0/TCP =
10.0.0.2;branch=3Dz9hG4bK83749</FONT>
<BR><FONT SIZE=3D2>3&nbsp;&nbsp; From: =
&lt;sip:alice@acme.com&gt;;tag=3Dabcd</FONT>
<BR><FONT SIZE=3D2>4&nbsp;&nbsp; To: &lt;sip:bob@acme.com&gt;</FONT>
<BR><FONT SIZE=3D2>5&nbsp;&nbsp; Call-ID: 1@10.0.0.2</FONT>
<BR><FONT SIZE=3D2>6&nbsp;&nbsp; CSeq: 2343 INVITE</FONT>
<BR><FONT SIZE=3D2>7&nbsp;&nbsp; Content-Type: message/external-body; =
access-type=3D&quot;URL&quot;;</FONT>
<BR><FONT =
SIZE=3D2>8&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; expiration=3D&quot;Mon, 24 June 2002 =
09:00:00 GMT&quot;;</FONT>
<BR><FONT =
SIZE=3D2>9&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; URL=3D&quot;<A =
HREF=3D"http://www.bogus.com/the-indirect-content" =
TARGET=3D"_blank">http://www.bogus.com/the-indirect-content</A>&quot;</F=
ONT>
<BR><FONT SIZE=3D2>10&nbsp; Content-Length: ...</FONT>
<BR><FONT SIZE=3D2>11&nbsp; &lt;CRLF&gt;</FONT>
<BR><FONT SIZE=3D2>12&nbsp; Content-Type: application/sdp</FONT>
<BR><FONT SIZE=3D2>13&nbsp; &lt;CRLF&gt;</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Line 12 is the start of the payload for this =
INVITE.</FONT>
<BR><FONT SIZE=3D2>Lines 7-9 define the content type of this payload =
(indirect content</FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
available at the given URL)</FONT>
</P>

<P><FONT SIZE=3D2>Line 12 gives the content type of the indirect =
content. It is actually</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; an entity =
header inside the payload (an encapsulated entity)</FONT>
</P>

<P><FONT SIZE=3D2>Line 13 marks the end of the entity headers for this =
encapsulated entity</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; and also =
marks the end of the message</FONT>
</P>

<P><FONT SIZE=3D2>This is consistent with the example given in section =
5.2.3 of RFC2046.</FONT>
<BR><FONT SIZE=3D2>I'm using MIME in the broadest sense defined in =
RFC2045/2046 to allow</FONT>
<BR><FONT SIZE=3D2>nesting or encapsulation of entities. This looks a =
bit strange compared</FONT>
<BR><FONT SIZE=3D2>to other SIP messages, but I believe it is perfectly =
valid MIME. I</FONT>
<BR><FONT SIZE=3D2>don't think we need to use message/sipfrag or even =
text/plain to</FONT>
<BR><FONT SIZE=3D2>accomplish</FONT>
<BR><FONT SIZE=3D2>this -- the capability to express nested entities =
and entity headers is</FONT>
<BR><FONT SIZE=3D2>part of MIME.</FONT>
<BR><FONT SIZE=3D2>I would be curious to hear from other MIME experts =
on this issue to see</FONT>
<BR><FONT SIZE=3D2>if I'm</FONT>
<BR><FONT SIZE=3D2>abusing something here.</FONT>
</P>

<P><FONT SIZE=3D2>/sean</FONT>
</P>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Sriram Parameswar [<A =
HREF=3D"mailto:sriramp@nortelnetworks.com">mailto:sriramp@nortelnetworks=
.com</A>] </FONT>
<BR><FONT SIZE=3D2>Sent: Sunday, July 07, 2002 6:35 PM</FONT>
<BR><FONT SIZE=3D2>To: Sean Olson; 'sip@ietf.org'</FONT>
<BR><FONT SIZE=3D2>Subject: =
draft-olson-sip-content-indirect-mech-00</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Sean, </FONT>
<BR><FONT SIZE=3D2>Well written draft!&nbsp; In Section 3.6 'Specifying =
the type of the indirect</FONT>
<BR><FONT SIZE=3D2>content' - </FONT>
<BR><FONT SIZE=3D2>it must be clearly noted that when including the =
content-type of the</FONT>
<BR><FONT SIZE=3D2>indirect content in the payload</FONT>
<BR><FONT SIZE=3D2>- there must be Content headers specifying that =
payload. Along with some</FONT>
<BR><FONT SIZE=3D2>textual description added to section 3.6,</FONT>
<BR><FONT SIZE=3D2>the example in that section will change to look =
something like:</FONT>
</P>

<P><FONT SIZE=3D2>Content-Type: message/external-body; =
access-type=3D&quot;URL&quot;; expiration=3D&quot;Mon,</FONT>
<BR><FONT SIZE=3D2>24 June 2002 09:00:00 GMT&quot;;</FONT>
<BR><FONT SIZE=3D2>URL=3D&quot;<A =
HREF=3D"http://www.bogus.com/the-indirect-content" =
TARGET=3D"_blank">http://www.bogus.com/the-indirect-content</A>&quot; =
</FONT>
<BR><FONT SIZE=3D2>Content-Type: text/plain </FONT>
<BR><FONT SIZE=3D2>Content-Length: ... </FONT>
<BR><FONT SIZE=3D2>&lt;CRLF&gt; </FONT>
<BR><FONT SIZE=3D2>Content-Type: application/sdp </FONT>
<BR><FONT SIZE=3D2>&lt;CRLF&gt; </FONT>
</P>

<P><FONT SIZE=3D2>You may also want to add a small section on how this =
mechanism satisfies</FONT>
<BR><FONT SIZE=3D2>the requirements laid down in the requirements =
document</FONT>
<BR><FONT SIZE=3D2>(draft-ietf-sipping-content-indirect-00). Just as an =
after thought, the</FONT>
<BR><FONT SIZE=3D2>additional Content-Type may also be message/sipfrag =
instead of</FONT>
<BR><FONT SIZE=3D2>text/plain.</FONT>
</P>

<P><FONT SIZE=3D2>Regards, </FONT>
<BR><FONT SIZE=3D2>Sriram </FONT>
<BR><FONT SIZE=3D2>__________________________________________ </FONT>
<BR><FONT SIZE=3D2>Sriram =
Parameswar&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp; Phone: 972-685-8540 </FONT>
<BR><FONT SIZE=3D2>Interactive Multimedia Server (IMS) Fax: =
972-684-3986 </FONT>
<BR><FONT SIZE=3D2>Nortel Networks, Richardson USA&nbsp; Email: =
sriramp@nortelnetworks.com </FONT>
</P>

</BODY>
</HTML>
------_=_NextPart_001_01C22775.73498490--

_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Jul  9 14:57:40 2002
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Thread-Topic: draft-olson-sip-content-indirect-mech-00
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From: "Sean Olson" <seanol@windows.microsoft.com>
To: "Sriram Parameswar" <sriramp@nortelnetworks.com>, <sip@ietf.org>
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Subject: [Sip] RE: draft-olson-sip-content-indirect-mech-00
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Thanks for the comments! 

I will try to dissect my original example to explain what I was
shooting for. I've added line numbers, plus some surrounding context
to hopefully make things a little clearer.

1   INVITE sip:bob@acme.com SIP/2.0
2   Via: SIP/2.0/TCP 10.0.0.2;branch=z9hG4bK83749
3   From: <sip:alice@acme.com>;tag=abcd
4   To: <sip:bob@acme.com>
5   Call-ID: 1@10.0.0.2
6   CSeq: 2343 INVITE
7   Content-Type: message/external-body; access-type="URL";
8                 expiration="Mon, 24 June 2002 09:00:00 GMT";
9                 URL="http://www.bogus.com/the-indirect-content"
10  Content-Length: ...
11  <CRLF>
12  Content-Type: application/sdp
13  <CRLF>


Line 12 is the start of the payload for this INVITE.
Lines 7-9 define the content type of this payload (indirect content
          available at the given URL)

Line 12 gives the content type of the indirect content. It is actually
        an entity header inside the payload (an encapsulated entity)

Line 13 marks the end of the entity headers for this encapsulated entity
        and also marks the end of the message

This is consistent with the example given in section 5.2.3 of RFC2046.
I'm using MIME in the broadest sense defined in RFC2045/2046 to allow
nesting or encapsulation of entities. This looks a bit strange compared
to other SIP messages, but I believe it is perfectly valid MIME. I
don't think we need to use message/sipfrag or even text/plain to
accomplish
this -- the capability to express nested entities and entity headers is
part of MIME.
I would be curious to hear from other MIME experts on this issue to see
if I'm
abusing something here.

/sean

-----Original Message-----
From: Sriram Parameswar [mailto:sriramp@nortelnetworks.com] 
Sent: Sunday, July 07, 2002 6:35 PM
To: Sean Olson; 'sip@ietf.org'
Subject: draft-olson-sip-content-indirect-mech-00


Sean, 
Well written draft!  In Section 3.6 'Specifying the type of the indirect
content' - 
it must be clearly noted that when including the content-type of the
indirect content in the payload
- there must be Content headers specifying that payload. Along with some
textual description added to section 3.6,
the example in that section will change to look something like:

Content-Type: message/external-body; access-type="URL"; expiration="Mon,
24 June 2002 09:00:00 GMT";
URL="http://www.bogus.com/the-indirect-content" 
Content-Type: text/plain 
Content-Length: ... 
<CRLF> 
Content-Type: application/sdp 
<CRLF> 

You may also want to add a small section on how this mechanism satisfies
the requirements laid down in the requirements document
(draft-ietf-sipping-content-indirect-00). Just as an after thought, the
additional Content-Type may also be message/sipfrag instead of
text/plain.

Regards, 
Sriram 
__________________________________________ 
Sriram Parameswar              Phone: 972-685-8540 
Interactive Multimedia Server (IMS) Fax: 972-684-3986 
Nortel Networks, Richardson USA  Email: sriramp@nortelnetworks.com 


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Jul  9 15:12:33 2002
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Thread-Topic: draft-olson-sip-content-indirect-mech-00
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From: "Sean Olson" <seanol@windows.microsoft.com>
To: "Sriram Parameswar" <sriramp@nortelnetworks.com>, <sip@ietf.org>
X-OriginalArrivalTime: 09 Jul 2002 18:29:43.0454 (UTC) FILETIME=[984C0BE0:01C22776]
Subject: [Sip] RE: draft-olson-sip-content-indirect-mech-00
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This is a multi-part message in MIME format.

--------------InterScan_NT_MIME_Boundary
Content-Type: multipart/alternative;
	boundary="----_=_NextPart_001_01C22776.9824F081"

------_=_NextPart_001_01C22776.9824F081
Content-Type: text/plain;
	charset="us-ascii"
Content-Transfer-Encoding: quoted-printable

If you are suggesting that we might want to indirectly refer to=20
message/sipfrag or text/plain content, I completely agree. The
example you cite below is almost exactly what I have proposed.
I just happen to use application/sdp instead of text/html as
the (indirect) Content-Type:
=20
thanks
/sean

	-----Original Message-----
	From: Sriram Parameswar [mailto:sriramp@nortelnetworks.com]=20
	Sent: Tuesday, July 09, 2002 11:22 AM
	To: Sean Olson; sip@ietf.org
	Subject: RE: draft-olson-sip-content-indirect-mech-00
=09
=09

	Thanks Sean - I would definitely like to hear from the MIME
experts - but here is an example from RFC 2017 section 3 which is more
along what I suggested. I think we need to take a closer look because
the current draft does borrow heavily from RFC2017:

	<quote RFC2017>=20
	    Content-type: message/external-body; access-type=3DURL;=20
	                  URL=3D"http://www.foo.com/file"=20

	    Content-type: text/html=20
	    Content-Transfer-Encoding: binary=20

	    THIS IS NOT REALLY THE BODY!=20

	</quote RFC2017>=20

	Regards,=20

	Sriram=20

	__________________________________________=20
	Sriram Parameswar              Phone: 972-685-8540=20
	Interactive Multimedia Server (IMS) Fax: 972-684-3986=20
	Nortel Networks, Richardson USA  Email:
sriramp@nortelnetworks.com=20


	-----Original Message-----=20
	From: Sean Olson [mailto:seanol@windows.microsoft.com]=20
	Sent: Tuesday, July 09, 2002 12:38 PM=20
	To: Parameswar, Sriram [NGC:B615:EXCH]; sip@ietf.org=20
	Subject: RE: draft-olson-sip-content-indirect-mech-00=20


	Thanks for the comments!=20

	I will try to dissect my original example to explain what I was=20
	shooting for. I've added line numbers, plus some surrounding
context=20
	to hopefully make things a little clearer.=20

	1   INVITE sip:bob@acme.com SIP/2.0=20
	2   Via: SIP/2.0/TCP 10.0.0.2;branch=3Dz9hG4bK83749=20
	3   From: <sip:alice@acme.com>;tag=3Dabcd=20
	4   To: <sip:bob@acme.com>=20
	5   Call-ID: 1@10.0.0.2=20
	6   CSeq: 2343 INVITE=20
	7   Content-Type: message/external-body; access-type=3D"URL";=20
	8                 expiration=3D"Mon, 24 June 2002 09:00:00 GMT";=20
	9
URL=3D"http://www.bogus.com/the-indirect-content"=20
	10  Content-Length: ...=20
	11  <CRLF>=20
	12  Content-Type: application/sdp=20
	13  <CRLF>=20


	Line 12 is the start of the payload for this INVITE.=20
	Lines 7-9 define the content type of this payload (indirect
content=20
	          available at the given URL)=20

	Line 12 gives the content type of the indirect content. It is
actually=20
	        an entity header inside the payload (an encapsulated
entity)=20

	Line 13 marks the end of the entity headers for this
encapsulated entity=20
	        and also marks the end of the message=20

	This is consistent with the example given in section 5.2.3 of
RFC2046.=20
	I'm using MIME in the broadest sense defined in RFC2045/2046 to
allow=20
	nesting or encapsulation of entities. This looks a bit strange
compared=20
	to other SIP messages, but I believe it is perfectly valid MIME.
I=20
	don't think we need to use message/sipfrag or even text/plain to

	accomplish=20
	this -- the capability to express nested entities and entity
headers is=20
	part of MIME.=20
	I would be curious to hear from other MIME experts on this issue
to see=20
	if I'm=20
	abusing something here.=20

	/sean=20

	-----Original Message-----=20
	From: Sriram Parameswar [mailto:sriramp@nortelnetworks.com]=20
	Sent: Sunday, July 07, 2002 6:35 PM=20
	To: Sean Olson; 'sip@ietf.org'=20
	Subject: draft-olson-sip-content-indirect-mech-00=20


	Sean,=20
	Well written draft!  In Section 3.6 'Specifying the type of the
indirect=20
	content' -=20
	it must be clearly noted that when including the content-type of
the=20
	indirect content in the payload=20
	- there must be Content headers specifying that payload. Along
with some=20
	textual description added to section 3.6,=20
	the example in that section will change to look something like:=20

	Content-Type: message/external-body; access-type=3D"URL";
expiration=3D"Mon,=20
	24 June 2002 09:00:00 GMT";=20
	URL=3D"http://www.bogus.com/the-indirect-content"=20
	Content-Type: text/plain=20
	Content-Length: ...=20
	<CRLF>=20
	Content-Type: application/sdp=20
	<CRLF>=20

	You may also want to add a small section on how this mechanism
satisfies=20
	the requirements laid down in the requirements document=20
	(draft-ietf-sipping-content-indirect-00). Just as an after
thought, the=20
	additional Content-Type may also be message/sipfrag instead of=20
	text/plain.=20

	Regards,=20
	Sriram=20
	__________________________________________=20
	Sriram Parameswar              Phone: 972-685-8540=20
	Interactive Multimedia Server (IMS) Fax: 972-684-3986=20
	Nortel Networks, Richardson USA  Email:
sriramp@nortelnetworks.com=20


------_=_NextPart_001_01C22776.9824F081
Content-Type: text/html;
	charset="us-ascii"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>Message</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2800.1050" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D084532718-09072002><FONT face=3DArial color=3D#0000ff =
size=3D2>If you=20
are suggesting that we might want to indirectly refer to =
</FONT></SPAN></DIV>
<DIV><SPAN class=3D084532718-09072002><FONT face=3DArial color=3D#0000ff =

size=3D2>message/sipfrag or text/plain content, I completely agree.=20
The</FONT></SPAN></DIV>
<DIV><SPAN class=3D084532718-09072002><FONT face=3DArial color=3D#0000ff =

size=3D2>example you cite below is almost exactly what I have=20
proposed.</FONT></SPAN></DIV>
<DIV><SPAN class=3D084532718-09072002><FONT face=3DArial color=3D#0000ff =
size=3D2>I just=20
happen to use application/sdp instead of text/html =
as</FONT></SPAN></DIV>
<DIV><SPAN class=3D084532718-09072002><FONT face=3DArial color=3D#0000ff =
size=3D2>the=20
(indirect) Content-Type:</FONT></SPAN></DIV>
<DIV><SPAN class=3D084532718-09072002><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D084532718-09072002><FONT face=3DArial color=3D#0000ff =

size=3D2>thanks</FONT></SPAN></DIV>
<DIV><SPAN class=3D084532718-09072002><FONT face=3DArial color=3D#0000ff =

size=3D2>/sean</FONT></SPAN></DIV>
<BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
  <DIV></DIV>
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft><FONT=20
  face=3DTahoma size=3D2>-----Original Message-----<BR><B>From:</B> =
Sriram=20
  Parameswar [mailto:sriramp@nortelnetworks.com] <BR><B>Sent:</B> =
Tuesday, July=20
  09, 2002 11:22 AM<BR><B>To:</B> Sean Olson; =
sip@ietf.org<BR><B>Subject:</B>=20
  RE: draft-olson-sip-content-indirect-mech-00<BR><BR></FONT></DIV>
  <P><FONT size=3D2>Thanks Sean - I would definitely like to hear from =
the MIME=20
  experts - but here is an example from RFC 2017 section 3 which is more =
along=20
  what I suggested. I think we need to take a closer look because the =
current=20
  draft does borrow heavily from RFC2017:</FONT></P>
  <P><FONT size=3D2>&lt;quote RFC2017&gt;</FONT> <BR><FONT=20
  size=3D2>&nbsp;&nbsp;&nbsp; Content-type: message/external-body;=20
  access-type=3DURL;</FONT> <BR><FONT=20
  =
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  URL=3D"<A href=3D"http://www.foo.com/file"=20
  target=3D_blank>http://www.foo.com/file</A>"</FONT> </P>
  <P><FONT size=3D2>&nbsp;&nbsp;&nbsp; Content-type: text/html</FONT> =
<BR><FONT=20
  size=3D2>&nbsp;&nbsp;&nbsp; Content-Transfer-Encoding: binary</FONT> =
</P>
  <P><FONT size=3D2>&nbsp;&nbsp;&nbsp; THIS IS NOT REALLY THE =
BODY!</FONT> </P>
  <P><FONT size=3D2>&lt;/quote RFC2017&gt;</FONT> </P>
  <P><FONT size=3D2>Regards,</FONT> </P>
  <P><FONT size=3D2>Sriram</FONT> </P>
  <P><FONT size=3D2>__________________________________________</FONT> =
<BR><FONT=20
  size=3D2>Sriram=20
  =
Parameswar&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;=20
  Phone: 972-685-8540</FONT> <BR><FONT size=3D2>Interactive Multimedia =
Server=20
  (IMS) Fax: 972-684-3986</FONT> <BR><FONT size=3D2>Nortel Networks, =
Richardson=20
  USA&nbsp; Email: sriramp@nortelnetworks.com</FONT> </P><BR>
  <P><FONT size=3D2>-----Original Message-----</FONT> <BR><FONT =
size=3D2>From: Sean=20
  Olson [<A=20
  =
href=3D"mailto:seanol@windows.microsoft.com">mailto:seanol@windows.micros=
oft.com</A>]</FONT>=20
  <BR><FONT size=3D2>Sent: Tuesday, July 09, 2002 12:38 PM</FONT> =
<BR><FONT=20
  size=3D2>To: Parameswar, Sriram [NGC:B615:EXCH]; sip@ietf.org</FONT> =
<BR><FONT=20
  size=3D2>Subject: RE: draft-olson-sip-content-indirect-mech-00</FONT> =
</P><BR>
  <P><FONT size=3D2>Thanks for the comments! </FONT></P>
  <P><FONT size=3D2>I will try to dissect my original example to explain =
what I=20
  was</FONT> <BR><FONT size=3D2>shooting for. I've added line numbers, =
plus some=20
  surrounding context</FONT> <BR><FONT size=3D2>to hopefully make things =
a little=20
  clearer.</FONT> </P>
  <P><FONT size=3D2>1&nbsp;&nbsp; INVITE sip:bob@acme.com SIP/2.0</FONT> =
<BR><FONT=20
  size=3D2>2&nbsp;&nbsp; Via: SIP/2.0/TCP =
10.0.0.2;branch=3Dz9hG4bK83749</FONT>=20
  <BR><FONT size=3D2>3&nbsp;&nbsp; From:=20
  &lt;sip:alice@acme.com&gt;;tag=3Dabcd</FONT> <BR><FONT =
size=3D2>4&nbsp;&nbsp; To:=20
  &lt;sip:bob@acme.com&gt;</FONT> <BR><FONT size=3D2>5&nbsp;&nbsp; =
Call-ID:=20
  1@10.0.0.2</FONT> <BR><FONT size=3D2>6&nbsp;&nbsp; CSeq: 2343 =
INVITE</FONT>=20
  <BR><FONT size=3D2>7&nbsp;&nbsp; Content-Type: message/external-body;=20
  access-type=3D"URL";</FONT> <BR><FONT=20
  =
size=3D2>8&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  expiration=3D"Mon, 24 June 2002 09:00:00 GMT";</FONT> <BR><FONT=20
  =
size=3D2>9&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
  URL=3D"<A href=3D"http://www.bogus.com/the-indirect-content"=20
  target=3D_blank>http://www.bogus.com/the-indirect-content</A>"</FONT> =
<BR><FONT=20
  size=3D2>10&nbsp; Content-Length: ...</FONT> <BR><FONT =
size=3D2>11&nbsp;=20
  &lt;CRLF&gt;</FONT> <BR><FONT size=3D2>12&nbsp; Content-Type:=20
  application/sdp</FONT> <BR><FONT size=3D2>13&nbsp; &lt;CRLF&gt;</FONT> =
</P><BR>
  <P><FONT size=3D2>Line 12 is the start of the payload for this =
INVITE.</FONT>=20
  <BR><FONT size=3D2>Lines 7-9 define the content type of this payload =
(indirect=20
  content</FONT> <BR><FONT=20
  size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
available at the=20
  given URL)</FONT> </P>
  <P><FONT size=3D2>Line 12 gives the content type of the indirect =
content. It is=20
  actually</FONT> <BR><FONT =
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; an=20
  entity header inside the payload (an encapsulated entity)</FONT> </P>
  <P><FONT size=3D2>Line 13 marks the end of the entity headers for this =

  encapsulated entity</FONT> <BR><FONT=20
  size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; and also marks the =
end of=20
  the message</FONT> </P>
  <P><FONT size=3D2>This is consistent with the example given in section =
5.2.3 of=20
  RFC2046.</FONT> <BR><FONT size=3D2>I'm using MIME in the broadest =
sense defined=20
  in RFC2045/2046 to allow</FONT> <BR><FONT size=3D2>nesting or =
encapsulation of=20
  entities. This looks a bit strange compared</FONT> <BR><FONT =
size=3D2>to other=20
  SIP messages, but I believe it is perfectly valid MIME. I</FONT> =
<BR><FONT=20
  size=3D2>don't think we need to use message/sipfrag or even text/plain =
to</FONT>=20
  <BR><FONT size=3D2>accomplish</FONT> <BR><FONT size=3D2>this -- the =
capability to=20
  express nested entities and entity headers is</FONT> <BR><FONT =
size=3D2>part of=20
  MIME.</FONT> <BR><FONT size=3D2>I would be curious to hear from other =
MIME=20
  experts on this issue to see</FONT> <BR><FONT size=3D2>if I'm</FONT> =
<BR><FONT=20
  size=3D2>abusing something here.</FONT> </P>
  <P><FONT size=3D2>/sean</FONT> </P>
  <P><FONT size=3D2>-----Original Message-----</FONT> <BR><FONT =
size=3D2>From:=20
  Sriram Parameswar [<A=20
  =
href=3D"mailto:sriramp@nortelnetworks.com">mailto:sriramp@nortelnetworks.=
com</A>]=20
  </FONT><BR><FONT size=3D2>Sent: Sunday, July 07, 2002 6:35 PM</FONT> =
<BR><FONT=20
  size=3D2>To: Sean Olson; 'sip@ietf.org'</FONT> <BR><FONT =
size=3D2>Subject:=20
  draft-olson-sip-content-indirect-mech-00</FONT> </P><BR>
  <P><FONT size=3D2>Sean, </FONT><BR><FONT size=3D2>Well written =
draft!&nbsp; In=20
  Section 3.6 'Specifying the type of the indirect</FONT> <BR><FONT=20
  size=3D2>content' - </FONT><BR><FONT size=3D2>it must be clearly noted =
that when=20
  including the content-type of the</FONT> <BR><FONT size=3D2>indirect =
content in=20
  the payload</FONT> <BR><FONT size=3D2>- there must be Content headers =
specifying=20
  that payload. Along with some</FONT> <BR><FONT size=3D2>textual =
description=20
  added to section 3.6,</FONT> <BR><FONT size=3D2>the example in that =
section will=20
  change to look something like:</FONT> </P>
  <P><FONT size=3D2>Content-Type: message/external-body; =
access-type=3D"URL";=20
  expiration=3D"Mon,</FONT> <BR><FONT size=3D2>24 June 2002 09:00:00 =
GMT";</FONT>=20
  <BR><FONT size=3D2>URL=3D"<A =
href=3D"http://www.bogus.com/the-indirect-content"=20
  target=3D_blank>http://www.bogus.com/the-indirect-content</A>" =
</FONT><BR><FONT=20
  size=3D2>Content-Type: text/plain </FONT><BR><FONT =
size=3D2>Content-Length: ...=20
  </FONT><BR><FONT size=3D2>&lt;CRLF&gt; </FONT><BR><FONT =
size=3D2>Content-Type:=20
  application/sdp </FONT><BR><FONT size=3D2>&lt;CRLF&gt; </FONT></P>
  <P><FONT size=3D2>You may also want to add a small section on how this =
mechanism=20
  satisfies</FONT> <BR><FONT size=3D2>the requirements laid down in the=20
  requirements document</FONT> <BR><FONT=20
  size=3D2>(draft-ietf-sipping-content-indirect-00). Just as an after =
thought,=20
  the</FONT> <BR><FONT size=3D2>additional Content-Type may also be=20
  message/sipfrag instead of</FONT> <BR><FONT =
size=3D2>text/plain.</FONT> </P>
  <P><FONT size=3D2>Regards, </FONT><BR><FONT size=3D2>Sriram =
</FONT><BR><FONT=20
  size=3D2>__________________________________________ </FONT><BR><FONT=20
  size=3D2>Sriram=20
  =
Parameswar&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;=20
  Phone: 972-685-8540 </FONT><BR><FONT size=3D2>Interactive Multimedia =
Server=20
  (IMS) Fax: 972-684-3986 </FONT><BR><FONT size=3D2>Nortel Networks, =
Richardson=20
  USA&nbsp; Email: sriramp@nortelnetworks.com=20
</FONT></P></BLOCKQUOTE></BODY></HTML>
=00
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Thread-Topic: draft-olson-sip-content-indirect-mech-00
Thread-Index: AcIneDsUPxJ1jc13T42EPbxNMFnnzQAAKn2w
From: "Sean Olson" <seanol@windows.microsoft.com>
To: "Sriram Parameswar" <sriramp@nortelnetworks.com>
Cc: <sip@ietf.org>
X-OriginalArrivalTime: 09 Jul 2002 18:47:32.0084 (UTC) FILETIME=[15400B40:01C22779]
Subject: [Sip] RE: draft-olson-sip-content-indirect-mech-00
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	charset="us-ascii"
Content-Transfer-Encoding: quoted-printable

I think we are saying the same thing... my intent is to send two
Content-Type:
headers. One says you are using message/external-body (content
indirection).
The other states the content type of the indirect content
(application/sdp, text/html,
whatever)
=20
/sean

	-----Original Message-----
	From: Sriram Parameswar [mailto:sriramp@nortelnetworks.com]=20
	Sent: Tuesday, July 09, 2002 11:41 AM
	To: Sean Olson
	Cc: sip@ietf.org
	Subject: RE: draft-olson-sip-content-indirect-mech-00
=09
=09
	Sean,
	=20
	One more reason for pushing for the additional Content-Type is
Section 7.4.1 of RFC3261 says - "The Internet media type of the message
body MUST be given by the Content-Type header field." Otherwise how will
the UAS receiving the INVITE know how to parse the body? But maybe I
should wait till the MIME experts give us their verdict :)
	=20
	Thanks,
	=20
	Sriram

	__________________________________________=20
	Sriram Parameswar              Phone: 972-685-8540=20
	Interactive Multimedia Server (IMS) Fax: 972-684-3986=20
	Nortel Networks, Richardson USA  Email:
sriramp@nortelnetworks.com=20

		-----Original Message-----
		From: Sean Olson [mailto:seanol@windows.microsoft.com]
		Sent: Tuesday, July 09, 2002 1:30 PM
		To: Parameswar, Sriram [NGC:B615:EXCH]; sip@ietf.org
		Subject: RE: draft-olson-sip-content-indirect-mech-00
	=09
	=09
		If you are suggesting that we might want to indirectly
refer to=20
		message/sipfrag or text/plain content, I completely
agree. The
		example you cite below is almost exactly what I have
proposed.
		I just happen to use application/sdp instead of
text/html as
		the (indirect) Content-Type:
		=20
		thanks
		/sean

			-----Original Message-----
			From: Sriram Parameswar
[mailto:sriramp@nortelnetworks.com]=20
			Sent: Tuesday, July 09, 2002 11:22 AM
			To: Sean Olson; sip@ietf.org
			Subject: RE:
draft-olson-sip-content-indirect-mech-00
		=09
		=09

			Thanks Sean - I would definitely like to hear
from the MIME experts - but here is an example from RFC 2017 section 3
which is more along what I suggested. I think we need to take a closer
look because the current draft does borrow heavily from RFC2017:

			<quote RFC2017>=20
			    Content-type: message/external-body;
access-type=3DURL;=20
			                  URL=3D"http://www.foo.com/file"=20

			    Content-type: text/html=20
			    Content-Transfer-Encoding: binary=20

			    THIS IS NOT REALLY THE BODY!=20

			</quote RFC2017>=20

			Regards,=20

			Sriram=20

			__________________________________________=20
			Sriram Parameswar              Phone:
972-685-8540=20
			Interactive Multimedia Server (IMS) Fax:
972-684-3986=20
			Nortel Networks, Richardson USA  Email:
sriramp@nortelnetworks.com=20


			-----Original Message-----=20
			From: Sean Olson
[mailto:seanol@windows.microsoft.com]=20
			Sent: Tuesday, July 09, 2002 12:38 PM=20
			To: Parameswar, Sriram [NGC:B615:EXCH];
sip@ietf.org=20
			Subject: RE:
draft-olson-sip-content-indirect-mech-00=20


			Thanks for the comments!=20

			I will try to dissect my original example to
explain what I was=20
			shooting for. I've added line numbers, plus some
surrounding context=20
			to hopefully make things a little clearer.=20

			1   INVITE sip:bob@acme.com SIP/2.0=20
			2   Via: SIP/2.0/TCP
10.0.0.2;branch=3Dz9hG4bK83749=20
			3   From: <sip:alice@acme.com>;tag=3Dabcd=20
			4   To: <sip:bob@acme.com>=20
			5   Call-ID: 1@10.0.0.2=20
			6   CSeq: 2343 INVITE=20
			7   Content-Type: message/external-body;
access-type=3D"URL";=20
			8                 expiration=3D"Mon, 24 June 2002
09:00:00 GMT";=20
			9
URL=3D"http://www.bogus.com/the-indirect-content"=20
			10  Content-Length: ...=20
			11  <CRLF>=20
			12  Content-Type: application/sdp=20
			13  <CRLF>=20


			Line 12 is the start of the payload for this
INVITE.=20
			Lines 7-9 define the content type of this
payload (indirect content=20
			          available at the given URL)=20

			Line 12 gives the content type of the indirect
content. It is actually=20
			        an entity header inside the payload (an
encapsulated entity)=20

			Line 13 marks the end of the entity headers for
this encapsulated entity=20
			        and also marks the end of the message=20

			This is consistent with the example given in
section 5.2.3 of RFC2046.=20
			I'm using MIME in the broadest sense defined in
RFC2045/2046 to allow=20
			nesting or encapsulation of entities. This looks
a bit strange compared=20
			to other SIP messages, but I believe it is
perfectly valid MIME. I=20
			don't think we need to use message/sipfrag or
even text/plain to=20
			accomplish=20
			this -- the capability to express nested
entities and entity headers is=20
			part of MIME.=20
			I would be curious to hear from other MIME
experts on this issue to see=20
			if I'm=20
			abusing something here.=20

			/sean=20

			-----Original Message-----=20
			From: Sriram Parameswar
[mailto:sriramp@nortelnetworks.com]=20
			Sent: Sunday, July 07, 2002 6:35 PM=20
			To: Sean Olson; 'sip@ietf.org'=20
			Subject:
draft-olson-sip-content-indirect-mech-00=20


			Sean,=20
			Well written draft!  In Section 3.6 'Specifying
the type of the indirect=20
			content' -=20
			it must be clearly noted that when including the
content-type of the=20
			indirect content in the payload=20
			- there must be Content headers specifying that
payload. Along with some=20
			textual description added to section 3.6,=20
			the example in that section will change to look
something like:=20

			Content-Type: message/external-body;
access-type=3D"URL"; expiration=3D"Mon,=20
			24 June 2002 09:00:00 GMT";=20
			URL=3D"http://www.bogus.com/the-indirect-content"=20
			Content-Type: text/plain=20
			Content-Length: ...=20
			<CRLF>=20
			Content-Type: application/sdp=20
			<CRLF>=20

			You may also want to add a small section on how
this mechanism satisfies=20
			the requirements laid down in the requirements
document=20
			(draft-ietf-sipping-content-indirect-00). Just
as an after thought, the=20
			additional Content-Type may also be
message/sipfrag instead of=20
			text/plain.=20

			Regards,=20
			Sriram=20
			__________________________________________=20
			Sriram Parameswar              Phone:
972-685-8540=20
			Interactive Multimedia Server (IMS) Fax:
972-684-3986=20
			Nortel Networks, Richardson USA  Email:
sriramp@nortelnetworks.com=20


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Content-Type: text/html;
	charset="us-ascii"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>Message</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2800.1050" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D274114618-09072002><FONT face=3DArial color=3D#0000ff =
size=3D2>I=20
think we are saying the same thing... my intent is to send two=20
Content-Type:</FONT></SPAN></DIV>
<DIV><SPAN class=3D274114618-09072002><FONT face=3DArial color=3D#0000ff =

size=3D2>headers. One says you are using message/external-body (content=20
indirection).</FONT></SPAN></DIV>
<DIV><SPAN class=3D274114618-09072002><FONT face=3DArial color=3D#0000ff =
size=3D2>The=20
other states the content type of the indirect content (application/sdp,=20
text/html,</FONT></SPAN></DIV>
<DIV><SPAN class=3D274114618-09072002><FONT face=3DArial color=3D#0000ff =

size=3D2>whatever)</FONT></SPAN></DIV>
<DIV><SPAN class=3D274114618-09072002><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D274114618-09072002><FONT face=3DArial color=3D#0000ff =

size=3D2>/sean</FONT></SPAN></DIV>
<BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
  <DIV></DIV>
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft><FONT=20
  face=3DTahoma size=3D2>-----Original Message-----<BR><B>From:</B> =
Sriram=20
  Parameswar [mailto:sriramp@nortelnetworks.com] <BR><B>Sent:</B> =
Tuesday, July=20
  09, 2002 11:41 AM<BR><B>To:</B> Sean Olson<BR><B>Cc:</B>=20
  sip@ietf.org<BR><B>Subject:</B> RE:=20
  draft-olson-sip-content-indirect-mech-00<BR><BR></FONT></DIV>
  <DIV><SPAN class=3D878473618-09072002><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2>Sean,</FONT></SPAN></DIV>
  <DIV><SPAN class=3D878473618-09072002><FONT face=3DArial =
color=3D#0000ff=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D878473618-09072002><FONT face=3DArial =
color=3D#0000ff size=3D2>One=20
  more reason for pushing for the additional Content-Type is =
</FONT></SPAN><SPAN=20
  class=3D878473618-09072002><FONT face=3DArial size=3D2><FONT =
color=3D#0000ff>Section=20
  7.4.1 of RFC3261 says&nbsp;- "</FONT><FONT face=3D"Times New Roman" =
size=3D3><FONT=20
  color=3D#0000ff>The Internet media type of the message body =
</FONT></FONT><FONT=20
  color=3D#0000ff><FONT face=3D"Times New Roman" size=3D1>MUST =
</FONT><FONT=20
  size=3D3><FONT face=3D"Times New Roman">be given by the </FONT><FONT=20
  face=3DArial>Content-Type </FONT><FONT face=3D"Times New Roman">header =
field.<SPAN=20
  class=3D878473618-09072002>" Otherwise how will the UAS receiving the =
INVITE=20
  know how to parse the body? But maybe I should wait till the MIME =
experts give=20
  us their verdict :)</SPAN></FONT></FONT></FONT></FONT></SPAN></DIV>
  <DIV><SPAN class=3D878473618-09072002><FONT face=3DArial =
size=3D2><FONT=20
  color=3D#0000ff><FONT size=3D3><FONT face=3D"Times New Roman"><SPAN=20
  =
class=3D878473618-09072002></SPAN></FONT></FONT></FONT></FONT></SPAN>&nbs=
p;</DIV>
  <DIV><SPAN class=3D878473618-09072002><FONT face=3DArial =
size=3D2><FONT=20
  color=3D#0000ff><FONT size=3D3><FONT face=3D"Times New Roman"><SPAN=20
  =
class=3D878473618-09072002>Thanks,</SPAN></FONT></FONT></FONT></FONT></SP=
AN></DIV>
  <DIV><SPAN class=3D878473618-09072002><FONT face=3DArial =
size=3D2><FONT=20
  color=3D#0000ff><FONT size=3D3><FONT face=3D"Times New Roman"><SPAN=20
  =
class=3D878473618-09072002></SPAN></FONT></FONT></FONT></FONT></SPAN>&nbs=
p;</DIV>
  <DIV><SPAN class=3D878473618-09072002><FONT face=3DArial =
size=3D2><FONT=20
  color=3D#0000ff><FONT size=3D3><FONT face=3D"Times New Roman"><SPAN=20
  =
class=3D878473618-09072002>Sriram</SPAN></FONT></FONT></FONT></FONT></SPA=
N></DIV>
  <P><B><FONT face=3D"Comic Sans MS"=20
  size=3D2>__________________________________________</FONT></B> =
<BR><B><FONT=20
  face=3D"Comic Sans MS" size=3D2>Sriram=20
  =
Parameswar&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;</FONT></B>=20
  <FONT face=3DArial size=3D2>Phone: 972-685-8540</FONT> <BR><FONT =
face=3DArial=20
  size=3D2>Interactive Multimedia Server (IMS) Fax: 972-684-3986</FONT> =
<BR><FONT=20
  face=3DArial size=3D2>Nortel Networks, Richardson USA&nbsp;</FONT> =
<FONT=20
  face=3D"Arial Narrow" size=3D2>Email: =
sriramp@nortelnetworks.com</FONT> </P>
  <BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
    <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
    size=3D2>-----Original Message-----<BR><B>From:</B> Sean Olson=20
    [mailto:seanol@windows.microsoft.com]<BR><B>Sent:</B> Tuesday, July =
09, 2002=20
    1:30 PM<BR><B>To:</B> Parameswar, Sriram [NGC:B615:EXCH];=20
    sip@ietf.org<BR><B>Subject:</B> RE:=20
    draft-olson-sip-content-indirect-mech-00<BR><BR></FONT></DIV>
    <DIV><SPAN class=3D084532718-09072002><FONT face=3DArial =
color=3D#0000ff size=3D2>If=20
    you are suggesting that we might want to indirectly refer to=20
    </FONT></SPAN></DIV>
    <DIV><SPAN class=3D084532718-09072002><FONT face=3DArial =
color=3D#0000ff=20
    size=3D2>message/sipfrag or text/plain content, I completely agree.=20
    The</FONT></SPAN></DIV>
    <DIV><SPAN class=3D084532718-09072002><FONT face=3DArial =
color=3D#0000ff=20
    size=3D2>example you cite below is almost exactly what I have=20
    proposed.</FONT></SPAN></DIV>
    <DIV><SPAN class=3D084532718-09072002><FONT face=3DArial =
color=3D#0000ff size=3D2>I=20
    just happen to use application/sdp instead of text/html=20
    as</FONT></SPAN></DIV>
    <DIV><SPAN class=3D084532718-09072002><FONT face=3DArial =
color=3D#0000ff=20
    size=3D2>the (indirect) Content-Type:</FONT></SPAN></DIV>
    <DIV><SPAN class=3D084532718-09072002><FONT face=3DArial =
color=3D#0000ff=20
    size=3D2></FONT></SPAN>&nbsp;</DIV>
    <DIV><SPAN class=3D084532718-09072002><FONT face=3DArial =
color=3D#0000ff=20
    size=3D2>thanks</FONT></SPAN></DIV>
    <DIV><SPAN class=3D084532718-09072002><FONT face=3DArial =
color=3D#0000ff=20
    size=3D2>/sean</FONT></SPAN></DIV>
    <BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
      <DIV></DIV>
      <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft><FONT=20
      face=3DTahoma size=3D2>-----Original Message-----<BR><B>From:</B> =
Sriram=20
      Parameswar [mailto:sriramp@nortelnetworks.com] <BR><B>Sent:</B> =
Tuesday,=20
      July 09, 2002 11:22 AM<BR><B>To:</B> Sean Olson;=20
      sip@ietf.org<BR><B>Subject:</B> RE:=20
      draft-olson-sip-content-indirect-mech-00<BR><BR></FONT></DIV>
      <P><FONT size=3D2>Thanks Sean - I would definitely like to hear =
from the=20
      MIME experts - but here is an example from RFC 2017 section 3 =
which is=20
      more along what I suggested. I think we need to take a closer look =
because=20
      the current draft does borrow heavily from RFC2017:</FONT></P>
      <P><FONT size=3D2>&lt;quote RFC2017&gt;</FONT> <BR><FONT=20
      size=3D2>&nbsp;&nbsp;&nbsp; Content-type: message/external-body;=20
      access-type=3DURL;</FONT> <BR><FONT=20
      =
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
      URL=3D"<A href=3D"http://www.foo.com/file"=20
      target=3D_blank>http://www.foo.com/file</A>"</FONT> </P>
      <P><FONT size=3D2>&nbsp;&nbsp;&nbsp; Content-type: =
text/html</FONT>=20
      <BR><FONT size=3D2>&nbsp;&nbsp;&nbsp; Content-Transfer-Encoding:=20
      binary</FONT> </P>
      <P><FONT size=3D2>&nbsp;&nbsp;&nbsp; THIS IS NOT REALLY THE =
BODY!</FONT>=20
</P>
      <P><FONT size=3D2>&lt;/quote RFC2017&gt;</FONT> </P>
      <P><FONT size=3D2>Regards,</FONT> </P>
      <P><FONT size=3D2>Sriram</FONT> </P>
      <P><FONT =
size=3D2>__________________________________________</FONT>=20
      <BR><FONT size=3D2>Sriram=20
      =
Parameswar&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;=20
      Phone: 972-685-8540</FONT> <BR><FONT size=3D2>Interactive =
Multimedia Server=20
      (IMS) Fax: 972-684-3986</FONT> <BR><FONT size=3D2>Nortel Networks, =

      Richardson USA&nbsp; Email: sriramp@nortelnetworks.com</FONT> =
</P><BR>
      <P><FONT size=3D2>-----Original Message-----</FONT> <BR><FONT =
size=3D2>From:=20
      Sean Olson [<A=20
      =
href=3D"mailto:seanol@windows.microsoft.com">mailto:seanol@windows.micros=
oft.com</A>]</FONT>=20
      <BR><FONT size=3D2>Sent: Tuesday, July 09, 2002 12:38 PM</FONT> =
<BR><FONT=20
      size=3D2>To: Parameswar, Sriram [NGC:B615:EXCH]; =
sip@ietf.org</FONT>=20
      <BR><FONT size=3D2>Subject: RE:=20
      draft-olson-sip-content-indirect-mech-00</FONT> </P><BR>
      <P><FONT size=3D2>Thanks for the comments! </FONT></P>
      <P><FONT size=3D2>I will try to dissect my original example to =
explain what=20
      I was</FONT> <BR><FONT size=3D2>shooting for. I've added line =
numbers, plus=20
      some surrounding context</FONT> <BR><FONT size=3D2>to hopefully =
make things=20
      a little clearer.</FONT> </P>
      <P><FONT size=3D2>1&nbsp;&nbsp; INVITE sip:bob@acme.com =
SIP/2.0</FONT>=20
      <BR><FONT size=3D2>2&nbsp;&nbsp; Via: SIP/2.0/TCP=20
      10.0.0.2;branch=3Dz9hG4bK83749</FONT> <BR><FONT =
size=3D2>3&nbsp;&nbsp; From:=20
      &lt;sip:alice@acme.com&gt;;tag=3Dabcd</FONT> <BR><FONT =
size=3D2>4&nbsp;&nbsp;=20
      To: &lt;sip:bob@acme.com&gt;</FONT> <BR><FONT =
size=3D2>5&nbsp;&nbsp;=20
      Call-ID: 1@10.0.0.2</FONT> <BR><FONT size=3D2>6&nbsp;&nbsp; CSeq: =
2343=20
      INVITE</FONT> <BR><FONT size=3D2>7&nbsp;&nbsp; Content-Type:=20
      message/external-body; access-type=3D"URL";</FONT> <BR><FONT=20
      =
size=3D2>8&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
      expiration=3D"Mon, 24 June 2002 09:00:00 GMT";</FONT> <BR><FONT=20
      =
size=3D2>9&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
      URL=3D"<A href=3D"http://www.bogus.com/the-indirect-content"=20
      =
target=3D_blank>http://www.bogus.com/the-indirect-content</A>"</FONT>=20
      <BR><FONT size=3D2>10&nbsp; Content-Length: ...</FONT> <BR><FONT=20
      size=3D2>11&nbsp; &lt;CRLF&gt;</FONT> <BR><FONT size=3D2>12&nbsp;=20
      Content-Type: application/sdp</FONT> <BR><FONT size=3D2>13&nbsp;=20
      &lt;CRLF&gt;</FONT> </P><BR>
      <P><FONT size=3D2>Line 12 is the start of the payload for this=20
      INVITE.</FONT> <BR><FONT size=3D2>Lines 7-9 define the content =
type of this=20
      payload (indirect content</FONT> <BR><FONT=20
      size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
available at=20
      the given URL)</FONT> </P>
      <P><FONT size=3D2>Line 12 gives the content type of the indirect =
content. It=20
      is actually</FONT> <BR><FONT=20
      size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; an entity =
header inside=20
      the payload (an encapsulated entity)</FONT> </P>
      <P><FONT size=3D2>Line 13 marks the end of the entity headers for =
this=20
      encapsulated entity</FONT> <BR><FONT=20
      size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; and also marks =
the end=20
      of the message</FONT> </P>
      <P><FONT size=3D2>This is consistent with the example given in =
section 5.2.3=20
      of RFC2046.</FONT> <BR><FONT size=3D2>I'm using MIME in the =
broadest sense=20
      defined in RFC2045/2046 to allow</FONT> <BR><FONT size=3D2>nesting =
or=20
      encapsulation of entities. This looks a bit strange =
compared</FONT>=20
      <BR><FONT size=3D2>to other SIP messages, but I believe it is =
perfectly=20
      valid MIME. I</FONT> <BR><FONT size=3D2>don't think we need to use =

      message/sipfrag or even text/plain to</FONT> <BR><FONT=20
      size=3D2>accomplish</FONT> <BR><FONT size=3D2>this -- the =
capability to=20
      express nested entities and entity headers is</FONT> <BR><FONT =
size=3D2>part=20
      of MIME.</FONT> <BR><FONT size=3D2>I would be curious to hear from =
other=20
      MIME experts on this issue to see</FONT> <BR><FONT size=3D2>if =
I'm</FONT>=20
      <BR><FONT size=3D2>abusing something here.</FONT> </P>
      <P><FONT size=3D2>/sean</FONT> </P>
      <P><FONT size=3D2>-----Original Message-----</FONT> <BR><FONT =
size=3D2>From:=20
      Sriram Parameswar [<A=20
      =
href=3D"mailto:sriramp@nortelnetworks.com">mailto:sriramp@nortelnetworks.=
com</A>]=20
      </FONT><BR><FONT size=3D2>Sent: Sunday, July 07, 2002 6:35 =
PM</FONT>=20
      <BR><FONT size=3D2>To: Sean Olson; 'sip@ietf.org'</FONT> <BR><FONT =

      size=3D2>Subject: draft-olson-sip-content-indirect-mech-00</FONT> =
</P><BR>
      <P><FONT size=3D2>Sean, </FONT><BR><FONT size=3D2>Well written =
draft!&nbsp; In=20
      Section 3.6 'Specifying the type of the indirect</FONT> <BR><FONT=20
      size=3D2>content' - </FONT><BR><FONT size=3D2>it must be clearly =
noted that=20
      when including the content-type of the</FONT> <BR><FONT =
size=3D2>indirect=20
      content in the payload</FONT> <BR><FONT size=3D2>- there must be =
Content=20
      headers specifying that payload. Along with some</FONT> <BR><FONT=20
      size=3D2>textual description added to section 3.6,</FONT> =
<BR><FONT=20
      size=3D2>the example in that section will change to look something =

      like:</FONT> </P>
      <P><FONT size=3D2>Content-Type: message/external-body; =
access-type=3D"URL";=20
      expiration=3D"Mon,</FONT> <BR><FONT size=3D2>24 June 2002 09:00:00 =

      GMT";</FONT> <BR><FONT size=3D2>URL=3D"<A=20
      href=3D"http://www.bogus.com/the-indirect-content"=20
      target=3D_blank>http://www.bogus.com/the-indirect-content</A>"=20
      </FONT><BR><FONT size=3D2>Content-Type: text/plain =
</FONT><BR><FONT=20
      size=3D2>Content-Length: ... </FONT><BR><FONT =
size=3D2>&lt;CRLF&gt;=20
      </FONT><BR><FONT size=3D2>Content-Type: application/sdp =
</FONT><BR><FONT=20
      size=3D2>&lt;CRLF&gt; </FONT></P>
      <P><FONT size=3D2>You may also want to add a small section on how =
this=20
      mechanism satisfies</FONT> <BR><FONT size=3D2>the requirements =
laid down in=20
      the requirements document</FONT> <BR><FONT=20
      size=3D2>(draft-ietf-sipping-content-indirect-00). Just as an =
after thought,=20
      the</FONT> <BR><FONT size=3D2>additional Content-Type may also be=20
      message/sipfrag instead of</FONT> <BR><FONT =
size=3D2>text/plain.</FONT> </P>
      <P><FONT size=3D2>Regards, </FONT><BR><FONT size=3D2>Sriram =
</FONT><BR><FONT=20
      size=3D2>__________________________________________ =
</FONT><BR><FONT=20
      size=3D2>Sriram=20
      =
Parameswar&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;=20
      Phone: 972-685-8540 </FONT><BR><FONT size=3D2>Interactive =
Multimedia Server=20
      (IMS) Fax: 972-684-3986 </FONT><BR><FONT size=3D2>Nortel Networks, =

      Richardson USA&nbsp; Email: sriramp@nortelnetworks.com=20
    </FONT></P></BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE></BODY></HTML>
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From: "Sriram Parameswar" <sriramp@nortelnetworks.com>
To: "'Sean Olson'" <seanol@windows.microsoft.com>
Cc: sip@ietf.org
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Subject: [Sip] RE: draft-olson-sip-content-indirect-mech-00
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Sean,
 
One more reason for pushing for the additional Content-Type is Section 7.4.1
of RFC3261 says - "The Internet media type of the message body MUST be given
by the Content-Type header field." Otherwise how will the UAS receiving the
INVITE know how to parse the body? But maybe I should wait till the MIME
experts give us their verdict :)
 
Thanks,
 
Sriram

__________________________________________ 
Sriram Parameswar              Phone: 972-685-8540 
Interactive Multimedia Server (IMS) Fax: 972-684-3986 
Nortel Networks, Richardson USA  Email: sriramp@nortelnetworks.com 

-----Original Message-----
From: Sean Olson [mailto:seanol@windows.microsoft.com]
Sent: Tuesday, July 09, 2002 1:30 PM
To: Parameswar, Sriram [NGC:B615:EXCH]; sip@ietf.org
Subject: RE: draft-olson-sip-content-indirect-mech-00


If you are suggesting that we might want to indirectly refer to 
message/sipfrag or text/plain content, I completely agree. The
example you cite below is almost exactly what I have proposed.
I just happen to use application/sdp instead of text/html as
the (indirect) Content-Type:
 
thanks
/sean

-----Original Message-----
From: Sriram Parameswar [mailto:sriramp@nortelnetworks.com] 
Sent: Tuesday, July 09, 2002 11:22 AM
To: Sean Olson; sip@ietf.org
Subject: RE: draft-olson-sip-content-indirect-mech-00



Thanks Sean - I would definitely like to hear from the MIME experts - but
here is an example from RFC 2017 section 3 which is more along what I
suggested. I think we need to take a closer look because the current draft
does borrow heavily from RFC2017:

<quote RFC2017> 
    Content-type: message/external-body; access-type=URL; 
                  URL=" http://www.foo.com/file <http://www.foo.com/file> " 

    Content-type: text/html 
    Content-Transfer-Encoding: binary 

    THIS IS NOT REALLY THE BODY! 

</quote RFC2017> 

Regards, 

Sriram 

__________________________________________ 
Sriram Parameswar              Phone: 972-685-8540 
Interactive Multimedia Server (IMS) Fax: 972-684-3986 
Nortel Networks, Richardson USA  Email: sriramp@nortelnetworks.com 


-----Original Message----- 
From: Sean Olson [ mailto:seanol@windows.microsoft.com
<mailto:seanol@windows.microsoft.com> ] 
Sent: Tuesday, July 09, 2002 12:38 PM 
To: Parameswar, Sriram [NGC:B615:EXCH]; sip@ietf.org 
Subject: RE: draft-olson-sip-content-indirect-mech-00 


Thanks for the comments! 

I will try to dissect my original example to explain what I was 
shooting for. I've added line numbers, plus some surrounding context 
to hopefully make things a little clearer. 

1   INVITE sip:bob@acme.com SIP/2.0 
2   Via: SIP/2.0/TCP 10.0.0.2;branch=z9hG4bK83749 
3   From: <sip:alice@acme.com>;tag=abcd 
4   To: <sip:bob@acme.com> 
5   Call-ID: 1@10.0.0.2 
6   CSeq: 2343 INVITE 
7   Content-Type: message/external-body; access-type="URL"; 
8                 expiration="Mon, 24 June 2002 09:00:00 GMT"; 
9                 URL=" http://www.bogus.com/the-indirect-content
<http://www.bogus.com/the-indirect-content> " 
10  Content-Length: ... 
11  <CRLF> 
12  Content-Type: application/sdp 
13  <CRLF> 


Line 12 is the start of the payload for this INVITE. 
Lines 7-9 define the content type of this payload (indirect content 
          available at the given URL) 

Line 12 gives the content type of the indirect content. It is actually 
        an entity header inside the payload (an encapsulated entity) 

Line 13 marks the end of the entity headers for this encapsulated entity 
        and also marks the end of the message 

This is consistent with the example given in section 5.2.3 of RFC2046. 
I'm using MIME in the broadest sense defined in RFC2045/2046 to allow 
nesting or encapsulation of entities. This looks a bit strange compared 
to other SIP messages, but I believe it is perfectly valid MIME. I 
don't think we need to use message/sipfrag or even text/plain to 
accomplish 
this -- the capability to express nested entities and entity headers is 
part of MIME. 
I would be curious to hear from other MIME experts on this issue to see 
if I'm 
abusing something here. 

/sean 

-----Original Message----- 
From: Sriram Parameswar [ mailto:sriramp@nortelnetworks.com
<mailto:sriramp@nortelnetworks.com> ] 
Sent: Sunday, July 07, 2002 6:35 PM 
To: Sean Olson; 'sip@ietf.org' 
Subject: draft-olson-sip-content-indirect-mech-00 


Sean, 
Well written draft!  In Section 3.6 'Specifying the type of the indirect 
content' - 
it must be clearly noted that when including the content-type of the 
indirect content in the payload 
- there must be Content headers specifying that payload. Along with some 
textual description added to section 3.6, 
the example in that section will change to look something like: 

Content-Type: message/external-body; access-type="URL"; expiration="Mon, 
24 June 2002 09:00:00 GMT"; 
URL=" http://www.bogus.com/the-indirect-content
<http://www.bogus.com/the-indirect-content> " 
Content-Type: text/plain 
Content-Length: ... 
<CRLF> 
Content-Type: application/sdp 
<CRLF> 

You may also want to add a small section on how this mechanism satisfies 
the requirements laid down in the requirements document 
(draft-ietf-sipping-content-indirect-00). Just as an after thought, the 
additional Content-Type may also be message/sipfrag instead of 
text/plain. 

Regards, 
Sriram 
__________________________________________ 
Sriram Parameswar              Phone: 972-685-8540 
Interactive Multimedia Server (IMS) Fax: 972-684-3986 
Nortel Networks, Richardson USA  Email: sriramp@nortelnetworks.com 


------_=_NextPart_001_01C22778.3597BA10
Content-Type: text/html;
	charset="iso-8859-1"

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=iso-8859-1">
<TITLE>Message</TITLE>

<META content="MSHTML 6.00.2713.1100" name=GENERATOR></HEAD>
<BODY>
<DIV><SPAN class=878473618-09072002><FONT face=Arial color=#0000ff 
size=2>Sean,</FONT></SPAN></DIV>
<DIV><SPAN class=878473618-09072002><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=878473618-09072002><FONT face=Arial color=#0000ff size=2>One 
more reason for pushing for the additional Content-Type is </FONT></SPAN><SPAN 
class=878473618-09072002><FONT face=Arial size=2><FONT color=#0000ff>Section 
7.4.1 of RFC3261 says&nbsp;- "</FONT><FONT face="Times New Roman" size=3><FONT 
color=#0000ff>The Internet media type of the message body </FONT></FONT><FONT 
color=#0000ff><FONT face="Times New Roman" size=1>MUST </FONT><FONT size=3><FONT 
face="Times New Roman">be given by the </FONT><FONT face=Arial>Content-Type 
</FONT><FONT face="Times New Roman">header field.<SPAN 
class=878473618-09072002>" Otherwise how will the UAS receiving the INVITE know 
how to parse the body? But maybe I should wait till the MIME experts give us 
their verdict :)</SPAN></FONT></FONT></FONT></FONT></SPAN></DIV>
<DIV><SPAN class=878473618-09072002><FONT face=Arial size=2><FONT 
color=#0000ff><FONT size=3><FONT face="Times New Roman"><SPAN 
class=878473618-09072002></SPAN></FONT></FONT></FONT></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=878473618-09072002><FONT face=Arial size=2><FONT 
color=#0000ff><FONT size=3><FONT face="Times New Roman"><SPAN 
class=878473618-09072002>Thanks,</SPAN></FONT></FONT></FONT></FONT></SPAN></DIV>
<DIV><SPAN class=878473618-09072002><FONT face=Arial size=2><FONT 
color=#0000ff><FONT size=3><FONT face="Times New Roman"><SPAN 
class=878473618-09072002></SPAN></FONT></FONT></FONT></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=878473618-09072002><FONT face=Arial size=2><FONT 
color=#0000ff><FONT size=3><FONT face="Times New Roman"><SPAN 
class=878473618-09072002>Sriram</SPAN></FONT></FONT></FONT></FONT></SPAN></DIV>
<P><B><FONT face="Comic Sans MS" 
size=2>__________________________________________</FONT></B> <BR><B><FONT 
face="Comic Sans MS" size=2>Sriram 
Parameswar&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> 
<FONT face=Arial size=2>Phone: 972-685-8540</FONT> <BR><FONT face=Arial 
size=2>Interactive Multimedia Server (IMS) Fax: 972-684-3986</FONT> <BR><FONT 
face=Arial size=2>Nortel Networks, Richardson USA&nbsp;</FONT> <FONT 
face="Arial Narrow" size=2>Email: sriramp@nortelnetworks.com</FONT> </P>
<BLOCKQUOTE dir=ltr style="MARGIN-RIGHT: 0px">
  <DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> Sean Olson 
  [mailto:seanol@windows.microsoft.com]<BR><B>Sent:</B> Tuesday, July 09, 2002 
  1:30 PM<BR><B>To:</B> Parameswar, Sriram [NGC:B615:EXCH]; 
  sip@ietf.org<BR><B>Subject:</B> RE: 
  draft-olson-sip-content-indirect-mech-00<BR><BR></FONT></DIV>
  <DIV><SPAN class=084532718-09072002><FONT face=Arial color=#0000ff size=2>If 
  you are suggesting that we might want to indirectly refer to 
  </FONT></SPAN></DIV>
  <DIV><SPAN class=084532718-09072002><FONT face=Arial color=#0000ff 
  size=2>message/sipfrag or text/plain content, I completely agree. 
  The</FONT></SPAN></DIV>
  <DIV><SPAN class=084532718-09072002><FONT face=Arial color=#0000ff 
  size=2>example you cite below is almost exactly what I have 
  proposed.</FONT></SPAN></DIV>
  <DIV><SPAN class=084532718-09072002><FONT face=Arial color=#0000ff size=2>I 
  just happen to use application/sdp instead of text/html as</FONT></SPAN></DIV>
  <DIV><SPAN class=084532718-09072002><FONT face=Arial color=#0000ff size=2>the 
  (indirect) Content-Type:</FONT></SPAN></DIV>
  <DIV><SPAN class=084532718-09072002><FONT face=Arial color=#0000ff 
  size=2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=084532718-09072002><FONT face=Arial color=#0000ff 
  size=2>thanks</FONT></SPAN></DIV>
  <DIV><SPAN class=084532718-09072002><FONT face=Arial color=#0000ff 
  size=2>/sean</FONT></SPAN></DIV>
  <BLOCKQUOTE dir=ltr style="MARGIN-RIGHT: 0px">
    <DIV></DIV>
    <DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left><FONT 
    face=Tahoma size=2>-----Original Message-----<BR><B>From:</B> Sriram 
    Parameswar [mailto:sriramp@nortelnetworks.com] <BR><B>Sent:</B> Tuesday, 
    July 09, 2002 11:22 AM<BR><B>To:</B> Sean Olson; 
    sip@ietf.org<BR><B>Subject:</B> RE: 
    draft-olson-sip-content-indirect-mech-00<BR><BR></FONT></DIV>
    <P><FONT size=2>Thanks Sean - I would definitely like to hear from the MIME 
    experts - but here is an example from RFC 2017 section 3 which is more along 
    what I suggested. I think we need to take a closer look because the current 
    draft does borrow heavily from RFC2017:</FONT></P>
    <P><FONT size=2>&lt;quote RFC2017&gt;</FONT> <BR><FONT 
    size=2>&nbsp;&nbsp;&nbsp; Content-type: message/external-body; 
    access-type=URL;</FONT> <BR><FONT 
    size=2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
    URL="<A href="http://www.foo.com/file" 
    target=_blank>http://www.foo.com/file</A>"</FONT> </P>
    <P><FONT size=2>&nbsp;&nbsp;&nbsp; Content-type: text/html</FONT> <BR><FONT 
    size=2>&nbsp;&nbsp;&nbsp; Content-Transfer-Encoding: binary</FONT> </P>
    <P><FONT size=2>&nbsp;&nbsp;&nbsp; THIS IS NOT REALLY THE BODY!</FONT> </P>
    <P><FONT size=2>&lt;/quote RFC2017&gt;</FONT> </P>
    <P><FONT size=2>Regards,</FONT> </P>
    <P><FONT size=2>Sriram</FONT> </P>
    <P><FONT size=2>__________________________________________</FONT> <BR><FONT 
    size=2>Sriram 
    Parameswar&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
    Phone: 972-685-8540</FONT> <BR><FONT size=2>Interactive Multimedia Server 
    (IMS) Fax: 972-684-3986</FONT> <BR><FONT size=2>Nortel Networks, Richardson 
    USA&nbsp; Email: sriramp@nortelnetworks.com</FONT> </P><BR>
    <P><FONT size=2>-----Original Message-----</FONT> <BR><FONT size=2>From: 
    Sean Olson [<A 
    href="mailto:seanol@windows.microsoft.com">mailto:seanol@windows.microsoft.com</A>]</FONT> 
    <BR><FONT size=2>Sent: Tuesday, July 09, 2002 12:38 PM</FONT> <BR><FONT 
    size=2>To: Parameswar, Sriram [NGC:B615:EXCH]; sip@ietf.org</FONT> <BR><FONT 
    size=2>Subject: RE: draft-olson-sip-content-indirect-mech-00</FONT> </P><BR>
    <P><FONT size=2>Thanks for the comments! </FONT></P>
    <P><FONT size=2>I will try to dissect my original example to explain what I 
    was</FONT> <BR><FONT size=2>shooting for. I've added line numbers, plus some 
    surrounding context</FONT> <BR><FONT size=2>to hopefully make things a 
    little clearer.</FONT> </P>
    <P><FONT size=2>1&nbsp;&nbsp; INVITE sip:bob@acme.com SIP/2.0</FONT> 
    <BR><FONT size=2>2&nbsp;&nbsp; Via: SIP/2.0/TCP 
    10.0.0.2;branch=z9hG4bK83749</FONT> <BR><FONT size=2>3&nbsp;&nbsp; From: 
    &lt;sip:alice@acme.com&gt;;tag=abcd</FONT> <BR><FONT size=2>4&nbsp;&nbsp; 
    To: &lt;sip:bob@acme.com&gt;</FONT> <BR><FONT size=2>5&nbsp;&nbsp; Call-ID: 
    1@10.0.0.2</FONT> <BR><FONT size=2>6&nbsp;&nbsp; CSeq: 2343 INVITE</FONT> 
    <BR><FONT size=2>7&nbsp;&nbsp; Content-Type: message/external-body; 
    access-type="URL";</FONT> <BR><FONT 
    size=2>8&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
    expiration="Mon, 24 June 2002 09:00:00 GMT";</FONT> <BR><FONT 
    size=2>9&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
    URL="<A href="http://www.bogus.com/the-indirect-content" 
    target=_blank>http://www.bogus.com/the-indirect-content</A>"</FONT> 
    <BR><FONT size=2>10&nbsp; Content-Length: ...</FONT> <BR><FONT 
    size=2>11&nbsp; &lt;CRLF&gt;</FONT> <BR><FONT size=2>12&nbsp; Content-Type: 
    application/sdp</FONT> <BR><FONT size=2>13&nbsp; &lt;CRLF&gt;</FONT> 
</P><BR>
    <P><FONT size=2>Line 12 is the start of the payload for this INVITE.</FONT> 
    <BR><FONT size=2>Lines 7-9 define the content type of this payload (indirect 
    content</FONT> <BR><FONT 
    size=2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; available at 
    the given URL)</FONT> </P>
    <P><FONT size=2>Line 12 gives the content type of the indirect content. It 
    is actually</FONT> <BR><FONT 
    size=2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; an entity header inside 
    the payload (an encapsulated entity)</FONT> </P>
    <P><FONT size=2>Line 13 marks the end of the entity headers for this 
    encapsulated entity</FONT> <BR><FONT 
    size=2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; and also marks the end of 
    the message</FONT> </P>
    <P><FONT size=2>This is consistent with the example given in section 5.2.3 
    of RFC2046.</FONT> <BR><FONT size=2>I'm using MIME in the broadest sense 
    defined in RFC2045/2046 to allow</FONT> <BR><FONT size=2>nesting or 
    encapsulation of entities. This looks a bit strange compared</FONT> 
    <BR><FONT size=2>to other SIP messages, but I believe it is perfectly valid 
    MIME. I</FONT> <BR><FONT size=2>don't think we need to use message/sipfrag 
    or even text/plain to</FONT> <BR><FONT size=2>accomplish</FONT> <BR><FONT 
    size=2>this -- the capability to express nested entities and entity headers 
    is</FONT> <BR><FONT size=2>part of MIME.</FONT> <BR><FONT size=2>I would be 
    curious to hear from other MIME experts on this issue to see</FONT> 
    <BR><FONT size=2>if I'm</FONT> <BR><FONT size=2>abusing something 
    here.</FONT> </P>
    <P><FONT size=2>/sean</FONT> </P>
    <P><FONT size=2>-----Original Message-----</FONT> <BR><FONT size=2>From: 
    Sriram Parameswar [<A 
    href="mailto:sriramp@nortelnetworks.com">mailto:sriramp@nortelnetworks.com</A>] 
    </FONT><BR><FONT size=2>Sent: Sunday, July 07, 2002 6:35 PM</FONT> <BR><FONT 
    size=2>To: Sean Olson; 'sip@ietf.org'</FONT> <BR><FONT size=2>Subject: 
    draft-olson-sip-content-indirect-mech-00</FONT> </P><BR>
    <P><FONT size=2>Sean, </FONT><BR><FONT size=2>Well written draft!&nbsp; In 
    Section 3.6 'Specifying the type of the indirect</FONT> <BR><FONT 
    size=2>content' - </FONT><BR><FONT size=2>it must be clearly noted that when 
    including the content-type of the</FONT> <BR><FONT size=2>indirect content 
    in the payload</FONT> <BR><FONT size=2>- there must be Content headers 
    specifying that payload. Along with some</FONT> <BR><FONT size=2>textual 
    description added to section 3.6,</FONT> <BR><FONT size=2>the example in 
    that section will change to look something like:</FONT> </P>
    <P><FONT size=2>Content-Type: message/external-body; access-type="URL"; 
    expiration="Mon,</FONT> <BR><FONT size=2>24 June 2002 09:00:00 GMT";</FONT> 
    <BR><FONT size=2>URL="<A href="http://www.bogus.com/the-indirect-content" 
    target=_blank>http://www.bogus.com/the-indirect-content</A>" 
    </FONT><BR><FONT size=2>Content-Type: text/plain </FONT><BR><FONT 
    size=2>Content-Length: ... </FONT><BR><FONT size=2>&lt;CRLF&gt; 
    </FONT><BR><FONT size=2>Content-Type: application/sdp </FONT><BR><FONT 
    size=2>&lt;CRLF&gt; </FONT></P>
    <P><FONT size=2>You may also want to add a small section on how this 
    mechanism satisfies</FONT> <BR><FONT size=2>the requirements laid down in 
    the requirements document</FONT> <BR><FONT 
    size=2>(draft-ietf-sipping-content-indirect-00). Just as an after thought, 
    the</FONT> <BR><FONT size=2>additional Content-Type may also be 
    message/sipfrag instead of</FONT> <BR><FONT size=2>text/plain.</FONT> </P>
    <P><FONT size=2>Regards, </FONT><BR><FONT size=2>Sriram </FONT><BR><FONT 
    size=2>__________________________________________ </FONT><BR><FONT 
    size=2>Sriram 
    Parameswar&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
    Phone: 972-685-8540 </FONT><BR><FONT size=2>Interactive Multimedia Server 
    (IMS) Fax: 972-684-3986 </FONT><BR><FONT size=2>Nortel Networks, Richardson 
    USA&nbsp; Email: sriramp@nortelnetworks.com 
</FONT></P></BLOCKQUOTE></BLOCKQUOTE></BODY></HTML>

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From daemon@optimus.ietf.org  Tue Jul  9 15:24:39 2002
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From: Adam Roach <adam@dynamicsoft.com>
To: "'Sriram Parameswar'" <sriramp@nortelnetworks.com>,
        "'Sean Olson'"
	 <seanol@windows.microsoft.com>
Cc: sip@ietf.org
Subject: RE: [Sip] RE: draft-olson-sip-content-indirect-mech-00
Date: Tue, 9 Jul 2002 14:54:28 -0400 
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Sriram --

I think you're misunderstanding the text in RFC 2017 that you quoted.
Here's a simple demonstration you can use to help convince yourself
that Sean's proposal is consistent with RFC 2017:

How many "Content-Type" lines are in the RFC 2017 snippet you quoted?
Two. One (the actual message content type) specifies
message/external-body, and the other (which is actually in the
message body) specifies the content type of the external content.

How many "Content-Type" lines are in the example in Sean's document?
Two. One (the actual message content type) specifies
message/external-body, and the other (which is actually in the
message body) specifies the content type of the external content.

How many "Content-Type" lines are in your proposed example?
Three. The first specifies message/external-body, and
the third (which is actually in the message body) specifies
the content type of the external content. The second
specifies "text/plain" for no openly apparent reason,
since there is no text/plain content related to this
message in any way.

Can you see how your proposal is the inconsistent one?

/a

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From: "Sriram Parameswar" <sriramp@nortelnetworks.com>
To: "'Adam Roach'" <adam@dynamicsoft.com>,
        "'Sean Olson'"
	 <seanol@windows.microsoft.com>
Cc: "'sip@ietf.org'" <sip@ietf.org>
Subject: RE: [Sip] RE: draft-olson-sip-content-indirect-mech-00
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Adam:

Thanks for the explanation its clear - sorry for the confusion. 

Sean:
Last comment we probably need the Content-ID header in the example: RFC 2046
"The encapsulated headers in ALL "message/external-body" entities MUST
include a Content-ID header field to give a unique identifier by which to
reference the data."

Thanks,

Sriram

__________________________________________
Sriram Parameswar              Phone: 972-685-8540
Interactive Multimedia Server (IMS) Fax: 972-684-3986
Nortel Networks, Richardson USA  Email: sriramp@nortelnetworks.com


-----Original Message-----
From: Adam Roach [mailto:adam@dynamicsoft.com]
Sent: Tuesday, July 09, 2002 1:54 PM
To: Parameswar, Sriram [NGC:B615:EXCH]; 'Sean Olson'
Cc: sip@ietf.org
Subject: RE: [Sip] RE: draft-olson-sip-content-indirect-mech-00


Sriram --

I think you're misunderstanding the text in RFC 2017 that you quoted.
Here's a simple demonstration you can use to help convince yourself
that Sean's proposal is consistent with RFC 2017:

How many "Content-Type" lines are in the RFC 2017 snippet you quoted?
Two. One (the actual message content type) specifies
message/external-body, and the other (which is actually in the
message body) specifies the content type of the external content.

How many "Content-Type" lines are in the example in Sean's document?
Two. One (the actual message content type) specifies
message/external-body, and the other (which is actually in the
message body) specifies the content type of the external content.

How many "Content-Type" lines are in your proposed example?
Three. The first specifies message/external-body, and
the third (which is actually in the message body) specifies
the content type of the external content. The second
specifies "text/plain" for no openly apparent reason,
since there is no text/plain content related to this
message in any way.

Can you see how your proposal is the inconsistent one?

/a

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
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<TITLE>RE: [Sip] RE: draft-olson-sip-content-indirect-mech-00</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>Adam:</FONT>
</P>

<P><FONT SIZE=3D2>Thanks for the explanation its clear - sorry for the =
confusion. </FONT>
</P>

<P><FONT SIZE=3D2>Sean:</FONT>
<BR><FONT SIZE=3D2>Last comment we probably need the Content-ID header =
in the example: RFC 2046 &quot;The encapsulated headers in ALL =
&quot;message/external-body&quot; entities MUST include a Content-ID =
header field to give a unique identifier by which to reference the =
data.&quot;</FONT></P>

<P><FONT SIZE=3D2>Thanks,</FONT>
</P>

<P><FONT SIZE=3D2>Sriram</FONT>
</P>

<P><FONT SIZE=3D2>__________________________________________</FONT>
<BR><FONT SIZE=3D2>Sriram =
Parameswar&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp; Phone: 972-685-8540</FONT>
<BR><FONT SIZE=3D2>Interactive Multimedia Server (IMS) Fax: =
972-684-3986</FONT>
<BR><FONT SIZE=3D2>Nortel Networks, Richardson USA&nbsp; Email: =
sriramp@nortelnetworks.com</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Adam Roach [<A =
HREF=3D"mailto:adam@dynamicsoft.com">mailto:adam@dynamicsoft.com</A>]</F=
ONT>
<BR><FONT SIZE=3D2>Sent: Tuesday, July 09, 2002 1:54 PM</FONT>
<BR><FONT SIZE=3D2>To: Parameswar, Sriram [NGC:B615:EXCH]; 'Sean =
Olson'</FONT>
<BR><FONT SIZE=3D2>Cc: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>Subject: RE: [Sip] RE: =
draft-olson-sip-content-indirect-mech-00</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Sriram --</FONT>
</P>

<P><FONT SIZE=3D2>I think you're misunderstanding the text in RFC 2017 =
that you quoted.</FONT>
<BR><FONT SIZE=3D2>Here's a simple demonstration you can use to help =
convince yourself</FONT>
<BR><FONT SIZE=3D2>that Sean's proposal is consistent with RFC =
2017:</FONT>
</P>

<P><FONT SIZE=3D2>How many &quot;Content-Type&quot; lines are in the =
RFC 2017 snippet you quoted?</FONT>
<BR><FONT SIZE=3D2>Two. One (the actual message content type) =
specifies</FONT>
<BR><FONT SIZE=3D2>message/external-body, and the other (which is =
actually in the</FONT>
<BR><FONT SIZE=3D2>message body) specifies the content type of the =
external content.</FONT>
</P>

<P><FONT SIZE=3D2>How many &quot;Content-Type&quot; lines are in the =
example in Sean's document?</FONT>
<BR><FONT SIZE=3D2>Two. One (the actual message content type) =
specifies</FONT>
<BR><FONT SIZE=3D2>message/external-body, and the other (which is =
actually in the</FONT>
<BR><FONT SIZE=3D2>message body) specifies the content type of the =
external content.</FONT>
</P>

<P><FONT SIZE=3D2>How many &quot;Content-Type&quot; lines are in your =
proposed example?</FONT>
<BR><FONT SIZE=3D2>Three. The first specifies message/external-body, =
and</FONT>
<BR><FONT SIZE=3D2>the third (which is actually in the message body) =
specifies</FONT>
<BR><FONT SIZE=3D2>the content type of the external content. The =
second</FONT>
<BR><FONT SIZE=3D2>specifies &quot;text/plain&quot; for no openly =
apparent reason,</FONT>
<BR><FONT SIZE=3D2>since there is no text/plain content related to =
this</FONT>
<BR><FONT SIZE=3D2>message in any way.</FONT>
</P>

<P><FONT SIZE=3D2>Can you see how your proposal is the inconsistent =
one?</FONT>
</P>

<P><FONT SIZE=3D2>/a</FONT>
</P>

</BODY>
</HTML>
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Subject: RE: [Sip] RE: draft-olson-sip-content-indirect-mech-00
Date: Tue, 9 Jul 2002 12:34:24 -0700
Message-ID: <F66A04C29AD9034A8205949AD0C9010403FCEBCF@win-msg-02.wingroup.windeploy.ntdev.microsoft.com>
Thread-Topic: [Sip] RE: draft-olson-sip-content-indirect-mech-00
Thread-Index: AcIne8UzW6gX5O6sTAWbxdlmiu4QPQAA9F3A
From: "Sean Olson" <seanol@windows.microsoft.com>
To: "Sriram Parameswar" <sriramp@nortelnetworks.com>,
        "Adam Roach" <adam@dynamicsoft.com>
Cc: <sip@ietf.org>
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Thanks, I'll make the Content-ID a MUST in the -01 version.
/sean

	-----Original Message-----
	From: Sriram Parameswar [mailto:sriramp@nortelnetworks.com]=20
	Sent: Tuesday, July 09, 2002 12:07 PM
	To: 'Adam Roach'; Sean Olson
	Cc: 'sip@ietf.org'
	Subject: RE: [Sip] RE: draft-olson-sip-content-indirect-mech-00
=09
=09

	Adam:=20

	Thanks for the explanation its clear - sorry for the confusion.=20

	Sean:=20
	Last comment we probably need the Content-ID header in the
example: RFC 2046 "The encapsulated headers in ALL
"message/external-body" entities MUST include a Content-ID header field
to give a unique identifier by which to reference the data."

	Thanks,=20

	Sriram=20

	__________________________________________=20
	Sriram Parameswar              Phone: 972-685-8540=20
	Interactive Multimedia Server (IMS) Fax: 972-684-3986=20
	Nortel Networks, Richardson USA  Email:
sriramp@nortelnetworks.com=20


	-----Original Message-----=20
	From: Adam Roach [mailto:adam@dynamicsoft.com]=20
	Sent: Tuesday, July 09, 2002 1:54 PM=20
	To: Parameswar, Sriram [NGC:B615:EXCH]; 'Sean Olson'=20
	Cc: sip@ietf.org=20
	Subject: RE: [Sip] RE: draft-olson-sip-content-indirect-mech-00=20


	Sriram --=20

	I think you're misunderstanding the text in RFC 2017 that you
quoted.=20
	Here's a simple demonstration you can use to help convince
yourself=20
	that Sean's proposal is consistent with RFC 2017:=20

	How many "Content-Type" lines are in the RFC 2017 snippet you
quoted?=20
	Two. One (the actual message content type) specifies=20
	message/external-body, and the other (which is actually in the=20
	message body) specifies the content type of the external
content.=20

	How many "Content-Type" lines are in the example in Sean's
document?=20
	Two. One (the actual message content type) specifies=20
	message/external-body, and the other (which is actually in the=20
	message body) specifies the content type of the external
content.=20

	How many "Content-Type" lines are in your proposed example?=20
	Three. The first specifies message/external-body, and=20
	the third (which is actually in the message body) specifies=20
	the content type of the external content. The second=20
	specifies "text/plain" for no openly apparent reason,=20
	since there is no text/plain content related to this=20
	message in any way.=20

	Can you see how your proposal is the inconsistent one?=20

	/a=20


------_=_NextPart_001_01C2277F.A154F9D5
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	charset="us-ascii"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>Message</TITLE>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2800.1050" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D230063419-09072002><FONT face=3DArial color=3D#0000ff =

size=3D2>Thanks, I'll make the Content-ID a MUST in the -01=20
version.</FONT></SPAN></DIV>
<DIV><SPAN class=3D230063419-09072002><FONT face=3DArial color=3D#0000ff =

size=3D2>/sean</FONT></SPAN></DIV>
<BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
  <DIV></DIV>
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft><FONT=20
  face=3DTahoma size=3D2>-----Original Message-----<BR><B>From:</B> =
Sriram=20
  Parameswar [mailto:sriramp@nortelnetworks.com] <BR><B>Sent:</B> =
Tuesday, July=20
  09, 2002 12:07 PM<BR><B>To:</B> 'Adam Roach'; Sean Olson<BR><B>Cc:</B> =

  'sip@ietf.org'<BR><B>Subject:</B> RE: [Sip] RE:=20
  draft-olson-sip-content-indirect-mech-00<BR><BR></FONT></DIV>
  <P><FONT size=3D2>Adam:</FONT> </P>
  <P><FONT size=3D2>Thanks for the explanation its clear - sorry for the =

  confusion. </FONT></P>
  <P><FONT size=3D2>Sean:</FONT> <BR><FONT size=3D2>Last comment we =
probably need=20
  the Content-ID header in the example: RFC 2046 "The encapsulated =
headers in=20
  ALL "message/external-body" entities MUST include a Content-ID header =
field to=20
  give a unique identifier by which to reference the data."</FONT></P>
  <P><FONT size=3D2>Thanks,</FONT> </P>
  <P><FONT size=3D2>Sriram</FONT> </P>
  <P><FONT size=3D2>__________________________________________</FONT> =
<BR><FONT=20
  size=3D2>Sriram=20
  =
Parameswar&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;=20
  Phone: 972-685-8540</FONT> <BR><FONT size=3D2>Interactive Multimedia =
Server=20
  (IMS) Fax: 972-684-3986</FONT> <BR><FONT size=3D2>Nortel Networks, =
Richardson=20
  USA&nbsp; Email: sriramp@nortelnetworks.com</FONT> </P><BR>
  <P><FONT size=3D2>-----Original Message-----</FONT> <BR><FONT =
size=3D2>From: Adam=20
  Roach [<A=20
  =
href=3D"mailto:adam@dynamicsoft.com">mailto:adam@dynamicsoft.com</A>]</FO=
NT>=20
  <BR><FONT size=3D2>Sent: Tuesday, July 09, 2002 1:54 PM</FONT> =
<BR><FONT=20
  size=3D2>To: Parameswar, Sriram [NGC:B615:EXCH]; 'Sean Olson'</FONT> =
<BR><FONT=20
  size=3D2>Cc: sip@ietf.org</FONT> <BR><FONT size=3D2>Subject: RE: [Sip] =
RE:=20
  draft-olson-sip-content-indirect-mech-00</FONT> </P><BR>
  <P><FONT size=3D2>Sriram --</FONT> </P>
  <P><FONT size=3D2>I think you're misunderstanding the text in RFC 2017 =
that you=20
  quoted.</FONT> <BR><FONT size=3D2>Here's a simple demonstration you =
can use to=20
  help convince yourself</FONT> <BR><FONT size=3D2>that Sean's proposal =
is=20
  consistent with RFC 2017:</FONT> </P>
  <P><FONT size=3D2>How many "Content-Type" lines are in the RFC 2017 =
snippet you=20
  quoted?</FONT> <BR><FONT size=3D2>Two. One (the actual message content =
type)=20
  specifies</FONT> <BR><FONT size=3D2>message/external-body, and the =
other (which=20
  is actually in the</FONT> <BR><FONT size=3D2>message body) specifies =
the content=20
  type of the external content.</FONT> </P>
  <P><FONT size=3D2>How many "Content-Type" lines are in the example in =
Sean's=20
  document?</FONT> <BR><FONT size=3D2>Two. One (the actual message =
content type)=20
  specifies</FONT> <BR><FONT size=3D2>message/external-body, and the =
other (which=20
  is actually in the</FONT> <BR><FONT size=3D2>message body) specifies =
the content=20
  type of the external content.</FONT> </P>
  <P><FONT size=3D2>How many "Content-Type" lines are in your proposed=20
  example?</FONT> <BR><FONT size=3D2>Three. The first specifies=20
  message/external-body, and</FONT> <BR><FONT size=3D2>the third (which =
is=20
  actually in the message body) specifies</FONT> <BR><FONT size=3D2>the =
content=20
  type of the external content. The second</FONT> <BR><FONT =
size=3D2>specifies=20
  "text/plain" for no openly apparent reason,</FONT> <BR><FONT =
size=3D2>since=20
  there is no text/plain content related to this</FONT> <BR><FONT =
size=3D2>message=20
  in any way.</FONT> </P>
  <P><FONT size=3D2>Can you see how your proposal is the inconsistent =
one?</FONT>=20
  </P>
  <P><FONT size=3D2>/a</FONT> </P></BLOCKQUOTE></BODY></HTML>
=00
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From daemon@optimus.ietf.org  Tue Jul  9 17:20:17 2002
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From: "Dean Willis" <dwillis@dynamicsoft.com>
To: <sip@ietf.org>
Date: Tue, 9 Jul 2002 15:40:30 -0500
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Subject: [Sip] Should service route discover be standards track or p-header
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Should the SIP Service route Discovery draft be an informational
P-header or should it be standards track? Some have argued that it is
more general than usual for P-headers. It seems to me to be useful
whenever it is useful for a registrar to return routing infromation that
UAs may use when they want request routing services from the domain
served by that registrar.

Opinions?

See:

http://search.ietf.org/internet-drafts/draft-willis-sip-scvrtdisco-06.tx
t


--
Dean



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From daemon@optimus.ietf.org  Tue Jul  9 17:29:12 2002
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Date: Tue, 09 Jul 2002 16:58:48 -0400
From: Henning Schulzrinne <hgs@cs.columbia.edu>
Organization: Columbia University
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To: Dean Willis <dwillis@dynamicsoft.com>
CC: sip@ietf.org
Subject: Re: [Sip] Should service route discover be standards track or p-header
References: <042201c22788$dd63dca0$eb036e3f@TXDWILLIS2>
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No, I think it should stay a P- header. Then, we will hopefully discover 
that it is generally useful and thus show that scarlet-letter headers 
are a flawed idea. All the other poor P- headers will then look up to 
this one and get some hope in their purgatory that redemption is possible.

Dean Willis wrote:
> Should the SIP Service route Discovery draft be an informational
> P-header or should it be standards track? Some have argued that it is
> more general than usual for P-headers. It seems to me to be useful
> whenever it is useful for a registrar to return routing infromation that
> UAs may use when they want request routing services from the domain
> served by that registrar.
> 
> Opinions?
> 
> See:
> 
> http://search.ietf.org/internet-drafts/draft-willis-sip-scvrtdisco-06.tx
> t
> 
> 
> --
> Dean
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip



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From daemon@optimus.ietf.org  Tue Jul  9 17:54:01 2002
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Date: Tue, 9 Jul 2002 14:32:47 -0700 (PDT)
From: Sean Olson <seancolson@yahoo.com>
Subject: Re: [Sip] Should service route discover be standards track or p-header
To: Dean Willis <dwillis@dynamicsoft.com>, sip@ietf.org
In-Reply-To: <042201c22788$dd63dca0$eb036e3f@TXDWILLIS2>
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It seems that the applicability of the
"Service Route Discovery" draft would be the
same as the "Header Field for Registering
Non-Adjacent Contacts" draft. Is there a
reason one should be a P-header and the other
should not?

/sean

--- Dean Willis <dwillis@dynamicsoft.com> wrote:
> 
> Should the SIP Service route Discovery draft be an
> informational
> P-header or should it be standards track? Some have
> argued that it is
> more general than usual for P-headers. It seems to
> me to be useful
> whenever it is useful for a registrar to return
> routing infromation that
> UAs may use when they want request routing services
> from the domain
> served by that registrar.
> 
> Opinions?
> 
> See:
> 
>
http://search.ietf.org/internet-drafts/draft-willis-sip-scvrtdisco-06.tx
> t
> 
> 
> --
> Dean
> 
> 
> 
> _______________________________________________
> Sip mailing list 
> https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP
> Protocol
> Use sip-implementors@cs.columbia.edu for questions
> on current sip
> Use sipping@ietf.org for new developments on the
> application of sip


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From daemon@optimus.ietf.org  Tue Jul  9 18:26:03 2002
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From: Adam Roach <adam@dynamicsoft.com>
To: "'Henning Schulzrinne'" <hgs@cs.columbia.edu>,
        Dean Willis
	 <dwillis@dynamicsoft.com>
Cc: sip@ietf.org
Subject: RE: [Sip] Should service route discover be standards track or p-h
	eader
Date: Tue, 9 Jul 2002 18:01:35 -0400 
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I'd argue that Service Route Discovery is generally useful,
and doesn't involve the working-group reservations that
the currently proposed P-headers do. Consider the
three possible criteria for classifying something as a
P header:

- Is it Proprietary? Certainly not -- it's undergoing
  working group standardization, and is generally
  appliable to SIP networks in general (not just specific
  networks with assumptions that don't hold in the
  Internet at large).

- Is it Private? No, it is going to be a published RFC.

- Is it Preliminary? I don't think so. If it is, we need to
  keep working on the draft until it's fully baked.

While I agree with Henning that the entire P-header ghetto
concept is complete rubbish, I don't think it would be
appropriate to crucify Service Route Discovery merely
to prove that point.

I'd prefer to continue to explain to the P-header proponents 
that "fire is hot and should not be touched" than to set up
a tripwire designed to throw them (and the rest of us) into
the campfire.

/a

> -----Original Message-----
> From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
> Sent: Tuesday, July 09, 2002 15:59
> To: Dean Willis
> Cc: sip@ietf.org
> Subject: Re: [Sip] Should service route discover be standards track or
> p-header
> 
> 
> No, I think it should stay a P- header. Then, we will 
> hopefully discover 
> that it is generally useful and thus show that scarlet-letter headers 
> are a flawed idea. All the other poor P- headers will then look up to 
> this one and get some hope in their purgatory that redemption 
> is possible.
> 
> Dean Willis wrote:
> > Should the SIP Service route Discovery draft be an informational
> > P-header or should it be standards track? Some have argued 
> that it is
> > more general than usual for P-headers. It seems to me to be useful
> > whenever it is useful for a registrar to return routing 
> infromation that
> > UAs may use when they want request routing services from the domain
> > served by that registrar.
> > 
> > Opinions?
> > 
> > See:
> > 
> > 
http://search.ietf.org/internet-drafts/draft-willis-sip-scvrtdisco-06.tx
> t
> 
> 
> --
> Dean
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip



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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip

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This list is for NEW development of the core SIP Protocol
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From daemon@optimus.ietf.org  Tue Jul  9 18:29:17 2002
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From: "Dean Willis" <dwillis@dynamicsoft.com>
To: <sip@ietf.org>, <agenda@ietf.org>
Date: Tue, 9 Jul 2002 17:02:33 -0500
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Subject: [Sip] Agenda, SIP WG, IETF 54
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The Siession Initiation Protocol (SIP) Working Group Agenda for IETF 54
is posted at:

http://www.softarmor.com/sipwg/meets/ietf54/agenda.html

Text follows. I will produce a hyperlinked version as time permits.

Agenda
------

Session 1 Monday July 15, 1530-1730, Room 301/302
1530 Agenda Bash
1535 Status and Charter, Chairs
1600 Refer open issues, Robert Sparks
1615 MIMEtypes open issues, Robert Sparks
1630 MESSAGE open issues, Ben Campbell
1645 Caller Preferences open issues, Jonathan Rosenberg
1700 SIP NAT open issues, Jonathan Rosenberg
1715 Digest Authentication enhancement open issues, James Undery
1730 Break





Session 2 Wednesday, July 17, 1300-1500, Room 301/302
1300 Agenda Bash
1305 Identity ppen issues, TBD (Watson, Jennings, Peterson?)
1330 URI Binding , TBD (Campbell/Schulzrinne?)
1345 SIP Compression open issues, Carsten Bormann
1400 Congestion Safety open issues, Dean Willis
1415 Discussion, referrals from SIPPING
      Content indirection, join, resource priority
1445 Future Work Planning
      Can we postpone cookie/state indefinitely?
1500 Break


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From daemon@optimus.ietf.org  Wed Jul 10 02:07:37 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
To: <sip@ietf.org>
Date: Wed, 10 Jul 2002 00:45:26 -0500
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Subject: [Sip] Cleanup on supplemental site "drafts" page
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I did some major cleanup at:

http://www.softarmor.com/sipwg/drafts/

Note that the "approved" documents are now in the rfced sub-page linked
at the top of this page.

If you've authored a draft that's supposed to be currently in the SIP
working group, please check here and make sure I'm tracking it. I wrote
all new data-entry forms for this stuff yesterday and re-entered a lot
of it this morning. I also "morgued" a lot of old documents, including
those things that have cleared the RFC editor queue and are now "RFC"s.

Also, please eyeball statuses and tracks and whatnot and let me know if
anything looks unreasonable.

Thanks,

--
Dean (your loyal E-3  Clerk/Typist, just call me Radar )


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From daemon@optimus.ietf.org  Wed Jul 10 06:01:02 2002
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Thread-Topic: ABNF for Replaces header
Thread-Index: AcIn9iwv5OZI2+wKQ6GEwjxbMJONvg==
From: "Ganesh Sivaraman" <ganesh.sivaraman@wipro.com>
To: <sip@ietf.org>
X-OriginalArrivalTime: 10 Jul 2002 09:35:22.0900 (UTC) FILETIME=[1D217D40:01C227F5]
Subject: [Sip] ABNF for Replaces header
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This is a multi-part message in MIME format.

------=_NextPartTM-000-c8a08143-c7ca-4fdf-a2b0-2165429b2d29
Content-Type: multipart/alternative;
	boundary="----_=_NextPart_001_01C227F5.1CEE7B64"

------_=_NextPart_001_01C227F5.1CEE7B64
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	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

Hi,
  Following is the ABNF for Replaces header as specified in =
draft-ietf-sip-replaces-02.txt:
=20
Replaces =3D "Replaces" HCOLON callid *(SEMI replaces-param)
replaces-param =3D to-tag / from-tag / generic-param
to-tag =3D "to-tag" EQUAL token
from-tag =3D "from-tag" EQUAL token
=20
According to this I could have Replaces headers like:
=20
Replaces: abcd@efgh;to-tag=3Dsdfsd --> (i)
Replaces: 1234ab@cde4 --> (ii)
=20
But the text right below the Replaces header's definition in the draft =
states the following:
=20
"A Replaces header MUST contain exactly one to-tag and exactly one =
from-tag, as they are required for unique dialog matching".
=20
According to this textual description the above examples ( i & ii) are =
incorrect.
=20
The ABNF doesn't seem to be matching this textual description. Is this a =
bug with the ABNF?
=20
Thanks & Regards
Ganesh
=20
=20
=20

------_=_NextPart_001_01C227F5.1CEE7B64
Content-Type: text/html;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<TITLE>Message</TITLE>

<META content=3D"MSHTML 5.00.3315.2870" name=3DGENERATOR></HEAD>
<BODY>
<DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D827060609-10072002>Hi,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D827060609-10072002>&nbsp; =
Following is=20
the&nbsp;ABNF for Replaces header as specified in=20
<STRONG>draft-ietf-sip-replaces-02.txt</STRONG>:</SPAN></FONT></DIV>
<DIV><SPAN class=3D827060609-10072002><FONT face=3DArial =
size=3D2>&nbsp;</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Replaces =3D "Replaces" HCOLON callid =
*(SEMI=20
replaces-param)</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>replaces-param =3D to-tag / from-tag /=20
generic-param</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>to-tag =3D "to-tag" EQUAL =
token</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>from-tag =3D "from-tag" EQUAL =
token</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN =
class=3D827060609-10072002>According to this I=20
could have&nbsp;Replaces header<SPAN class=3D852004009-10072002>s</SPAN> =

like:</SPAN></FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN =
class=3D827060609-10072002>Replaces:=20
abcd@efgh;to-tag=3Dsdfsd --&gt; (i)</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN =
class=3D827060609-10072002>Replaces:=20
1234ab@cde4 --&gt; (ii)</SPAN></FONT></DIV>
<DIV><SPAN class=3D827060609-10072002></SPAN><FONT face=3DArial=20
size=3D2>&nbsp;</FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D827060609-10072002>But =
the text right=20
below the Replaces header's definition in the draft states the=20
following:</SPAN></FONT></DIV>
<DIV><SPAN class=3D827060609-10072002></SPAN><FONT face=3DArial=20
size=3D2>&nbsp;</FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN =
class=3D827060609-10072002>"<STRONG>A Replaces=20
header MUST contain exactly one to-tag and exactly one from-tag, as they =
are=20
required for unique dialog matching</STRONG>".</SPAN></FONT></DIV>
<DIV><SPAN class=3D827060609-10072002></SPAN><FONT face=3DArial=20
size=3D2>&nbsp;</FONT></DIV>
<DIV><FONT size=3D2><FONT face=3DArial><SPAN =
class=3D827060609-10072002>According to=20
this textual description the above examples&nbsp;( i &amp; ii) are=20
incorrect<SPAN =
class=3D852004009-10072002>.</SPAN></SPAN></FONT></FONT></DIV>
<DIV><SPAN class=3D827060609-10072002></SPAN><FONT face=3DArial=20
size=3D2>&nbsp;</FONT></DIV>
<DIV><SPAN class=3D827060609-10072002><FONT face=3DArial size=3D2>The =
ABNF doesn't=20
seem to be matching this textual description. Is this a bug with the=20
ABNF?</FONT></SPAN></DIV>
<DIV><SPAN class=3D827060609-10072002></SPAN><FONT face=3DArial=20
size=3D2>&nbsp;</FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D827060609-10072002>Thanks =
&amp;=20
Regards</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D827060609-10072002>Ganesh</SPAN></FONT></DIV>
<DIV><SPAN class=3D827060609-10072002></SPAN><FONT face=3DArial=20
size=3D2>&nbsp;</FONT></DIV>
<DIV><SPAN class=3D827060609-10072002><FONT face=3DArial=20
size=3D2>&nbsp;</FONT></DIV></SPAN>
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From daemon@optimus.ietf.org  Wed Jul 10 07:49:18 2002
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Subject: RE: [Sip] ABNF for Replaces header
Date: Wed, 10 Jul 2002 06:16:12 -0500
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Thread-Topic: ABNF for Replaces header
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I don't see this so much as a bug in the ABNF as much as a failure in =
the language.  By specifying the ABNF as such, you can have:
=20
    Replaces: you@whereyoureat.com; to-tag=3Dthetotag; =
from-tag=3Dthefromtag
=20
or
=20
    Replaces: you@whereyoureat.com; from-tag=3Dthefromtag; =
to-tag=3Dthetotag
=20
or
=20
    Replaces: you@whereyoureat.com; myextension=3Dwhatever; =
from-tag=3Dthefromtag; anotherextension=3Dwhatelse; to-tag=3Dthetotag
=20
The ABNF can't specify what is mandatory without specifying the order in =
which it should occur.  I think specifying the ABNF as-is and then =
having the trailing comment is just a way of saying, I don't care what =
order you put the parameters in, but I do care that you have these two =
parameters.  So yes, your two examples below are both invalid.
=20
- rob

-----Original Message-----
From: Ganesh Sivaraman [mailto:ganesh.sivaraman@wipro.com]
Sent: Wednesday, July 10, 2002 4:35 AM
To: sip@ietf.org
Subject: [Sip] ABNF for Replaces header


Hi,
  Following is the ABNF for Replaces header as specified in =
draft-ietf-sip-replaces-02.txt:
=20
Replaces =3D "Replaces" HCOLON callid *(SEMI replaces-param)
replaces-param =3D to-tag / from-tag / generic-param
to-tag =3D "to-tag" EQUAL token
from-tag =3D "from-tag" EQUAL token
=20
According to this I could have Replaces headers like:
=20
Replaces: abcd@efgh;to-tag=3Dsdfsd --> (i)
Replaces: 1234ab@cde4 --> (ii)
=20
But the text right below the Replaces header's definition in the draft =
states the following:
=20
"A Replaces header MUST contain exactly one to-tag and exactly one =
from-tag, as they are required for unique dialog matching".
=20
According to this textual description the above examples ( i & ii) are =
incorrect.
=20
The ABNF doesn't seem to be matching this textual description. Is this a =
bug with the ABNF?
=20
Thanks & Regards
Ganesh
=20
=20
=20


------_=_NextPart_001_01C22803.32DA3034
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	charset="iso-8859-1"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<TITLE>Message</TITLE>

<META content=3D"MSHTML 6.00.2716.2200" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D872190811-10072002>I=20
don't see this so much as a bug in the ABNF as much as a failure in the=20
language.&nbsp; By specifying the ABNF as such, you can=20
have:</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D872190811-10072002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D872190811-10072002>&nbsp;&nbsp;&nbsp; Replaces: <A=20
href=3D"mailto:you@whereyoureat.com">you@whereyoureat.com</A>; =
to-tag=3Dthetotag;=20
from-tag=3Dthefromtag</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D872190811-10072002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D872190811-10072002>or</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D872190811-10072002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D872190811-10072002>&nbsp;&nbsp;&nbsp; Replaces: <A=20
href=3D"mailto:you@whereyoureat.com">you@whereyoureat.com</A>;=20
from-tag=3Dthefromtag; to-tag=3Dthetotag</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D872190811-10072002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D872190811-10072002>or</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D872190811-10072002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D872190811-10072002>&nbsp;&nbsp;&nbsp; Replaces: <A=20
href=3D"mailto:you@whereyoureat.com">you@whereyoureat.com</A>;=20
myextension=3Dwhatever; from-tag=3Dthefromtag; =
anotherextension=3Dwhatelse;=20
to-tag=3Dthetotag</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D872190811-10072002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D872190811-10072002>The=20
ABNF can't specify what is mandatory without specifying the order in =
which it=20
should occur.&nbsp; I think specifying the ABNF as-is and then having =
the=20
trailing comment is just a way of saying, I don't care what order you =
put the=20
parameters in, but I do care that you have these two parameters.&nbsp; =
So yes,=20
your two examples below are both invalid.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN=20
class=3D872190811-10072002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial color=3D#0000ff size=3D2><SPAN =
class=3D872190811-10072002>-=20
rob</SPAN></FONT></DIV>
<BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> Ganesh Sivaraman=20
  [mailto:ganesh.sivaraman@wipro.com]<BR><B>Sent:</B> Wednesday, July =
10, 2002=20
  4:35 AM<BR><B>To:</B> sip@ietf.org<BR><B>Subject:</B> [Sip] ABNF for =
Replaces=20
  header<BR><BR></FONT></DIV>
  <DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D827060609-10072002>Hi,</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN =
class=3D827060609-10072002>&nbsp; Following=20
  is the&nbsp;ABNF for Replaces header as specified in=20
  <STRONG>draft-ietf-sip-replaces-02.txt</STRONG>:</SPAN></FONT></DIV>
  <DIV><SPAN class=3D827060609-10072002><FONT face=3DArial=20
size=3D2></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2>Replaces =3D "Replaces" HCOLON callid =
*(SEMI=20
  replaces-param)</FONT></DIV>
  <DIV><FONT face=3DArial size=3D2>replaces-param =3D to-tag / from-tag =
/=20
  generic-param</FONT></DIV>
  <DIV><FONT face=3DArial size=3D2>to-tag =3D "to-tag" EQUAL =
token</FONT></DIV>
  <DIV><FONT face=3DArial size=3D2>from-tag =3D "from-tag" EQUAL =
token</FONT></DIV>
  <DIV>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN =
class=3D827060609-10072002>According to this=20
  I could have&nbsp;Replaces header<SPAN =
class=3D852004009-10072002>s</SPAN>=20
  like:</SPAN></FONT></DIV>
  <DIV>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN =
class=3D827060609-10072002>Replaces:=20
  abcd@efgh;to-tag=3Dsdfsd --&gt; (i)</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN =
class=3D827060609-10072002>Replaces:=20
  1234ab@cde4 --&gt; (ii)</SPAN></FONT></DIV>
  <DIV><SPAN class=3D827060609-10072002></SPAN><FONT face=3DArial=20
  size=3D2></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN class=3D827060609-10072002>But =
the text right=20
  below the Replaces header's definition in the draft states the=20
  following:</SPAN></FONT></DIV>
  <DIV><SPAN class=3D827060609-10072002></SPAN><FONT face=3DArial=20
  size=3D2></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN =
class=3D827060609-10072002>"<STRONG>A=20
  Replaces header MUST contain exactly one to-tag and exactly one =
from-tag, as=20
  they are required for unique dialog =
matching</STRONG>".</SPAN></FONT></DIV>
  <DIV><SPAN class=3D827060609-10072002></SPAN><FONT face=3DArial=20
  size=3D2></FONT>&nbsp;</DIV>
  <DIV><FONT size=3D2><FONT face=3DArial><SPAN =
class=3D827060609-10072002>According to=20
  this textual description the above examples&nbsp;( i &amp; ii) are=20
  incorrect<SPAN =
class=3D852004009-10072002>.</SPAN></SPAN></FONT></FONT></DIV>
  <DIV><SPAN class=3D827060609-10072002></SPAN><FONT face=3DArial=20
  size=3D2></FONT>&nbsp;</DIV>
  <DIV><SPAN class=3D827060609-10072002><FONT face=3DArial size=3D2>The =
ABNF doesn't=20
  seem to be matching this textual description. Is this a bug with the=20
  ABNF?</FONT></SPAN></DIV>
  <DIV><SPAN class=3D827060609-10072002></SPAN><FONT face=3DArial=20
  size=3D2></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN =
class=3D827060609-10072002>Thanks &amp;=20
  Regards</SPAN></FONT></DIV>
  <DIV><FONT face=3DArial size=3D2><SPAN=20
  class=3D827060609-10072002>Ganesh</SPAN></FONT></DIV>
  <DIV><SPAN class=3D827060609-10072002></SPAN><FONT face=3DArial=20
  size=3D2></FONT>&nbsp;</DIV>
  <DIV><SPAN class=3D827060609-10072002><FONT face=3DArial=20
  size=3D2></FONT>&nbsp;</DIV></SPAN>
  <DIV></SPAN><FONT face=3DArial=20
size=3D2></FONT>&nbsp;</DIV></DIV></BLOCKQUOTE></BODY></HTML>

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From daemon@optimus.ietf.org  Wed Jul 10 07:55:55 2002
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To: Rob Phillips <rob@netrake.com>
CC: sip@ietf.org
Subject: Re: [Sip] ABNF for Replaces header
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Hi,

I would say that it is NOT an error from an ABNF poinf of view if one or
more of those parameters are missing - ie you will not get a parser
error. But, it is a bug from an APPLICATION point of view, since the
applications does require all the parameters.

(Another example is the URI, where some parameters sometimes are
mandatory/forbidden depending on the SIP header the URI is defined for.
Again, it is not a ABNF bug, but an APPLICATION bug, if some parameter
is missing, or if some parameter is present even if it shouldn't.)

Of course it is possible to make the parameters mandatory also in the
ABNF, but in that case, which was said, you would have to also specify
the order of them.

...OR you can add comments in the ABNF saying that all the parameters
are mandatory. I don't know if that is "according to the rules", but at
least Megaco uses "at-most-once", "at-most-one" etc -type of comments in
the ABNF grammar.

Just my thoughts...

Regards,

Christer Holmberg
Ericsson Finland


Rob Phillips wrote:

>  I don't see this so much as a bug in the ABNF as much as a failure in
> the language.  By specifying the ABNF as such, you can have:
> Replaces: you@whereyoureat.com; to-tag=thetotag;
> from-tag=thefromtagor    Replaces: you@whereyoureat.com;
> from-tag=thefromtag; to-tag=thetotagor    Replaces:
> you@whereyoureat.com; myextension=whatever; from-tag=thefromtag;
> anotherextension=whatelse; to-tag=thetotagThe ABNF can't specify what
> is mandatory without specifying the order in which it should occur.  I
> think specifying the ABNF as-is and then having the trailing comment
> is just a way of saying, I don't care what order you put the
> parameters in, but I do care that you have these two parameters.  So
> yes, your two examples below are both invalid.- rob
>
>      -----Original Message-----
>      From: Ganesh Sivaraman [mailto:ganesh.sivaraman@wipro.com]
>      Sent: Wednesday, July 10, 2002 4:35 AM
>      To: sip@ietf.org
>      Subject: [Sip] ABNF for Replaces header
>
>      Hi,  Following is the ABNF for Replaces header as specified
>      in draft-ietf-sip-replaces-02.txt:Replaces = "Replaces"
>      HCOLON callid *(SEMI replaces-param)replaces-param = to-tag
>      / from-tag / generic-paramto-tag = "to-tag" EQUAL
>      tokenfrom-tag = "from-tag" EQUAL token According to this I
>      could have Replaces headers like: Replaces:
>      abcd@efgh;to-tag=sdfsd --> (i)Replaces: 1234ab@cde4 -->
>      (ii)But the text right below the Replaces header's
>      definition in the draft states the following:"A Replaces
>      header MUST contain exactly one to-tag and exactly one
>      from-tag, as they are required for unique dialog
>      matching".According to this textual description the above
>      examples ( i & ii) are incorrect.The ABNF doesn't seem to be
>      matching this textual description. Is this a bug with the
>      ABNF?Thanks & RegardsGanesh
>


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Spoofing is very simple without some form of authentication, merely
crafting a SIP packet with a an ethernet source IP address and
corresponding SIP information within the SIP/SDP body one could cause
many things to go wrong with an established call, ranging from denial of
service to redirection.


KANUMURI Sreeram wrote:
> 
> >
> > I don't think proxy needs to authenticate re-invite, as re-invite is used
> > for changing session parameters of an existing call
> 
> >>For this very reason you should reauthenticate...so that someone doesnt
> >>spoof a forged INVITE with bad parameters within it back to the client.
> 
> TRUE but can you tell me how can "someone  spoof a forged INVITE" ??
> like the call is established between the end users and it is olny session
> parameters change.
> As there is Already the authentication done.
> olny the UA in the call can change teh parameters with a new invite.....??
> 
> so can u make me  clearer??
> 
> Sreeram
> 
> >
> > Ranjit
> > ----- Original Message -----
> > From: ShavinderPal Singh <ShavinderPal.Singh@esamsungumit.com>
> > To: <sip@ietf.org>; Jonathan Rosemberg <jdrosen@dynamicsoft.com>
> > Sent: Monday, July 08, 2002 5:21 PM
> > Subject: [Sip] Authentication for re-INVITE.
> >
> > >
> > > Is a SIP proxy supposed to Authenticate a re-INVITE also? If no, then if
> > > some proxy in Authenticating and it asks for Auth Info from the UA, what
> > > is the UA supposed to do? Is it mandatory that it should send the Auth
> > > Info?
> > >
> > > Please suggest!
> > >
> > > Thanks & Regards
> > > s
> > >
> > > Ph.: 091-124-6383474 (e 250)
> > > __________________________________________
> > > I had a good grip on life, but then the handle broke.
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the application of sip
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Wed Jul 10 11:01:51 2002
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From: "Feldman, Michael" <mfeldman@bb.directv.com>
To: sip@ietf.org
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Subject: [Sip] SIP info method - rfc2976
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I know this is probably a little late, but....


I do not see why 
the info method is supposed to follow an established signaling path.  It
would be helpful (in my case at least) to send session information to a
third party destination not involved with the SIP call.



section 1 intro:
"It is necessary that the mid-session signaling information traverse the 
post session setup SIP signaling path.  This is the path taken by SIP
re-Invites, Byes, etc..."

it also mentions it in section 2...


thanks,
mike feldman
mfeldman@bb.directv.com

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From daemon@optimus.ietf.org  Wed Jul 10 11:25:19 2002
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From: Gordon Ledgard <gledgard@iperia.com>
To: "'Feldman, Michael'" <mfeldman@bb.directv.com>, sip@ietf.org
Subject: RE: [Sip] SIP info method - rfc2976
Date: Wed, 10 Jul 2002 10:53:40 -0400
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If INFO didn't follow the same rules as the rest of the SIP traffic,
it would require its own whole set of procedures to validate it and
make sure it belonged there.

Is your "third party destination" also a SIP endpoint? Why is it in
the conversation? Who says so? I believe what you are looking for can
be solved via a whole slew of other mechanisms besides SIP. 

And yep, you're a bit late. :)

Gordon Ledgard


-----Original Message-----
From: Feldman, Michael [mailto:mfeldman@bb.directv.com]
Sent: Wednesday, July 10, 2002 10:37 AM
To: sip@ietf.org
Subject: [Sip] SIP info method - rfc2976


I know this is probably a little late, but....


I do not see why 
the info method is supposed to follow an established signaling path.  It
would be helpful (in my case at least) to send session information to a
third party destination not involved with the SIP call.



section 1 intro:
"It is necessary that the mid-session signaling information traverse the 
post session setup SIP signaling path.  This is the path taken by SIP
re-Invites, Byes, etc..."

it also mentions it in section 2...


thanks,
mike feldman
mfeldman@bb.directv.com

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Wed Jul 10 12:30:41 2002
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From: Adam Roach <adam@dynamicsoft.com>
To: "'Feldman, Michael'" <mfeldman@bb.directv.com>, sip@ietf.org
Subject: RE: [Sip] SIP info method - rfc2976
Date: Wed, 10 Jul 2002 12:05:06 -0400
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The problem is that there is a widespread misconception that
you can do whatever the heck you want with INFO, and still
call what you're doing "SIP."

In all liklihood, you're trying to accomplish a task that
falls in one of the two categories:

a) Something the SIP working group has thought of, and
   has a solution to, or

b) Some useful application that we haven't thought of yet

So, instead of trying to overload INFO (which is looking like
an increasingly bad idea as time goes on), you'd probably be
better starting on sip-implementors (see the automatically-
inserted note at the end of this message), and describing your
application. If it falls into category (a) (and I suspect it
does -- we've been at this for a while, and have a ton of
tools), then someone can point you at the relevant RFCs/drafts.
If it falls into (b), then we can start working on the tools
you need to implement your service.

/a

> -----Original Message-----
> From: Feldman, Michael [mailto:mfeldman@bb.directv.com]
> Sent: Wednesday, July 10, 2002 9:37
> To: sip@ietf.org
> Subject: [Sip] SIP info method - rfc2976
> 
> 
> I know this is probably a little late, but....
> 
> 
> I do not see why 
> the info method is supposed to follow an established 
> signaling path.  It
> would be helpful (in my case at least) to send session 
> information to a
> third party destination not involved with the SIP call.
> 
> 
> 
> section 1 intro:
> "It is necessary that the mid-session signaling information 
> traverse the 
> post session setup SIP signaling path.  This is the path taken by SIP
> re-Invites, Byes, etc..."
> 
> it also mentions it in section 2...
> 
> 
> thanks,
> mike feldman
> mfeldman@bb.directv.com

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Wed Jul 10 16:02:45 2002
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Date: Wed, 10 Jul 2002 21:34:09 +0200
To: Ben Campbell <bcampbell@dynamicsoft.com>
From: Jiri Kuthan <kuthan@fokus.gmd.de>
Cc: sip@ietf.org
In-Reply-To: <3D29E679.7060405@dynamicsoft.com>
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 <5.1.0.14.0.20020703200900.00b3d110@iptel.org>
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Subject: [Sip] Re: MESSAGE advertising in REGISTERs
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What I actually do not understand is why we have not
put this requirement on all supported methods. Fixing
it for a specific methods seems better (though ugly)
to me than fixing it for none.

-Jiri

At 09:22 PM 7/8/2002, Ben Campbell wrote:
>I do not understand why we would expect support of MESSAGE to give a client a higher requirement for registration than for clients that do not support MESSAGE.
>
>Jiri Kuthan wrote:
>>Though advertising MESSAGE capability in REGISTERs does
>>not hurt in any scenario, not advertising hurts in some.
>>That's why I would much prefer having it as a MUST.
>>-Jiri
>>At 06:51 PM 7/3/2002, Ben Campbell wrote:
>>
>>>Certainly it will be needed to make many server related scenarios work. It is optional because MESSAGE can also be used in a pure end-to-end mode without assuming that such servers exist. For example, a MESSAGE UAC may discover the contact for a MESSAGE UAS from presence information (i.e. a contact in a presence doc), then send the MESSAGE request directly with no intervening servers. In this architecture, there is not necessarily a need to advertise support at a registrar.
>>>
>>>Jiri Kuthan wrote:
>>>
>>>>Can someone tell me why the MESSAGE I-D mandates advertising
>>>>of support in REGISTERs of a MESSAGE-capable UAC only *optionally*?
>>>>Making it optional gains little knowledge to a SIP server -- a server does not know anything it could reliably use in its processing logic.
>>>>Think of a SIP server which acts as a proxy server and message store.
>>>>If it knew, that downstream UAS was not capable of processing MESSAGEs,
>>>>it could immediately store an incoming MESSAGE, send 202 upstream,
>>>>and relay the message to first UAC which registers and advertises support
>>>>for MESSAGE in REGISTER.
>>>>Because the server does not know it, it has to relay unnecessarily a message to an upstream UAS, which will deny it. Then, whenever a REGISTER arrives, the server will try to dump message store to it,
>>>>(hoping a UAC capable of MESSAGE registered) resulting in quite
>>>>uneffective behaviour.
>>>>-Jiri
>


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From daemon@optimus.ietf.org  Wed Jul 10 16:39:40 2002
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Subject: Re: [Sip] REFER: Who has implemented REFER without creating
	subscriptions?
From: Robert Sparks <rsparks@dynamicsoft.com>
To: Robert Sparks <rsparks@dynamicsoft.com>
Cc: sip@ietf.org
In-Reply-To: <1025189183.2344.35.camel@dhcp150.dfw.dynamicsoft.com>
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I've received no responses indicating someone has implemented
accepting REFER and not sending NOTIFY. As sip-events matured,
the perceived potential need for that behavior has diminished.

So, I am changing REFER to require the implicit subscription,
so at least one NOTIFY will always be sent. (An implementation
that needs to be extremely lightweight can send just one NOTIFY
terminating the subscription). This change makes the 
Required/Supported mechanics discussed in 
draft-sparks-sip-refer-3265disc-00 unnecessary. I will remove
them.

When the submission blackout is listed, I will submit the
revised 3265disc draft as draft-ietf-sip-refer-06.

RjS

On Thu, 2002-06-27 at 09:46, Robert Sparks wrote:
> This was part of the last message I sent, but it was buried
> pretty deep.
> 
> When we made the decision to go with implicit subscriptions
> work around the finite length non-INVITE transaction, we made
> serving that subscription SHOULD strength to allow for implementations
> that couldn't deal with the weight of serving subscriptions.
> 
> Do the original proponents of allowing implementations to never
> send a notify still see this need? Are there any implementations
> out there that accept a REFER but don't send NOTIFYs?
> 
> The subscription concept has substantially matured (in particular,
> Subscription-State is explicitly communicated). As Dan points out
> in his last note, an implementation that needed to be very lightweight
> could send exactly one NOTIFY terminating the subscription. If
> serving the subscription (even if it involved sending just that
> one NOTIFY) were strictly mandatory, the error space is much smaller.
> 
> RjS
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip



_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Robert Sparks'" <rsparks@dynamicsoft.com>
Cc: sip@ietf.org
Subject: RE: [Sip] REFER: Who has implemented REFER without creating subsc
	riptions?
Date: Wed, 10 Jul 2002 17:21:16 -0400
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Of course we observe that "at least one" means some implementations
send one NOTIFY upon completion of REFER.  The new spec requires at
least two, an immediate one, and the one upon completion.

I agree we don't need the Required/Supported mechanism however.
All of us with "exactly one" will have to upgrade, but that's the
cost of implementing a draft.

Brian

> -----Original Message-----
> From: Robert Sparks [mailto:rsparks@dynamicsoft.com]
> Sent: Wednesday, July 10, 2002 4:15 PM
> To: Robert Sparks
> Cc: sip@ietf.org
> Subject: Re: [Sip] REFER: Who has implemented REFER without creating
> subscriptions?
> 
> 
> I've received no responses indicating someone has implemented
> accepting REFER and not sending NOTIFY. As sip-events matured,
> the perceived potential need for that behavior has diminished.
> 
> So, I am changing REFER to require the implicit subscription,
> so at least one NOTIFY will always be sent. (An implementation
> that needs to be extremely lightweight can send just one NOTIFY
> terminating the subscription). This change makes the 
> Required/Supported mechanics discussed in 
> draft-sparks-sip-refer-3265disc-00 unnecessary. I will remove
> them.
> 
> When the submission blackout is listed, I will submit the
> revised 3265disc draft as draft-ietf-sip-refer-06.
> 
> RjS
> 
> On Thu, 2002-06-27 at 09:46, Robert Sparks wrote:
> > This was part of the last message I sent, but it was buried
> > pretty deep.
> > 
> > When we made the decision to go with implicit subscriptions
> > work around the finite length non-INVITE transaction, we made
> > serving that subscription SHOULD strength to allow for 
> implementations
> > that couldn't deal with the weight of serving subscriptions.
> > 
> > Do the original proponents of allowing implementations to never
> > send a notify still see this need? Are there any implementations
> > out there that accept a REFER but don't send NOTIFYs?
> > 
> > The subscription concept has substantially matured (in particular,
> > Subscription-State is explicitly communicated). As Dan points out
> > in his last note, an implementation that needed to be very 
> lightweight
> > could send exactly one NOTIFY terminating the subscription. If
> > serving the subscription (even if it involved sending just that
> > one NOTIFY) were strictly mandatory, the error space is 
> much smaller.
> > 
> > RjS
> > 
> > 
> > 
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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From daemon@optimus.ietf.org  Wed Jul 10 17:55:35 2002
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        Robert Sparks
	 <rsparks@dynamicsoft.com>
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> -----Original Message-----
> From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
>
> The new [REFER] spec requires at least two [NOTIFYs],
> an immediate one, and the one upon completion.

As Dan pointed out and Robert reiterated, this isn't
precisely true. You can get away with just one
NOTIFY: the immediate one. As long as the "Subscription-
State" header indicates "terminated," this behavior
complies with the draft. The distinction is important,
since it doesn't require recipients of REFER to implement
the full event mechanism.

/a

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From daemon@optimus.ietf.org  Thu Jul 11 03:43:04 2002
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Date: Thu, 11 Jul 2002 10:10:46 +0300
From: Christer Holmberg <christer.holmberg@lmf.ericsson.se>
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Subject: [Sip] Re: [Sipping] draft-mahy-sipping-connect-reuse-reqs-00
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Hi,

I think it is important to think of who is allowed to "demand" what. Can we
always assume the client makes the final decission? What if the server - for
whatever reason - doens't want to/isn't able to reuse the connection? Shall
it send some kind of error response to the request where the "demand" is
specified, or shall it be allowed to use another, or establish a new,
connection? Shall the server be allowed to "demand" a re-use of connection,
even if the client didn't? How does this affect backward compability, if the
indicators aren't understood by all the nodes? Do we need
Supported/Require/Proxy-Require tags?

Regards,

Christer Holmberg
Ericsson Finland



> Good point. Is this is a request or a demand by the client to re-use
> the connection? I think we need the possibility for the client to
> *demand* that the connection be re-used. There may also be situations
> where re-use is just an optimization.
>
> /sean
>
> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@lmf.ericsson.se]
> Sent: Tuesday, July 09, 2002 9:41 AM
> To: Sean Olson; Gonzalo Camarillo; rohan@cisco.com; sip@ietf.org;
> sipping@ietf.org
> Subject: Re: [Sipping] draft-mahy-sipping-connect-reuse-reqs-00
>
> Another comment,
>
> I also think it is important to remember that there MAY be scenarios,
> especially if a connection is used for multiple sessions, where one
> wants to use separate connections per each direction - even if the
> client indicates he wants to use the same connection for the response.
> Just to think of before the MUSTs are added...
>
> Regards,
>
> Christer Holmberg
> Ericsson Finland
>
> Christer Holmberg wrote:
>
> > Hi Sean,
> >
> > > Is this really out of scope for this draft? Naturally
> > > the more re-use you make of the connection, the more benefit you get
>
> > > from re-use ;-)
> >
> > Of course, but in that case it should be mentioned in the draft :)
> >
> > The only thing the draft currently talks about the problem how to make
>
> > sure a single connection is used for both directions, per session -
> > which is a separate issue from reusing connections for multiple
> > sessions.
> >
> > And, since I don't think any draft currently talks about the reuse of
> > connections for multiple sessions (bis talks about keeping the
> > connection open to make sure the transaction uses the same
> > transaction), maybe it could be added to the connect-reuse draft?
> >
> > Just some thoughts...
> >
> > Regards,
> >
> > Christer Holmberg
> > Ericsson Finland
> >
> > >
> > >
> > > /sean
> > >
> > > -----Original Message-----
> > > From: Christer Holmberg [mailto:christer.holmberg@lmf.ericsson.se]
> > > Sent: Tuesday, July 09, 2002 6:30 AM
> > > To: Gonzalo Camarillo
> > > Cc: Sean Olson; rohan@cisco.com; sip@ietf.org; sipping@ietf.org
> > > Subject: Re: [Sipping] draft-mahy-sipping-connect-reuse-reqs-00
> > >
> > > Hi,
> > >
> > > RFC3261/bis-9 mainly talks about using a single connection (or, one
> > > in each
> > > direction) for a single transaction, and the connect-reuse draft
> > > talks about being able to use a single connection for messages in
> > > both direction, for a single session. Sometimes, however, when we
> > > talk about "reuse of connection" we mean when a single connection
> > > (or, one in each
> > > direction) is used for multiple sessions. I think it would be a good
> > > idea to mention that that kind of "reuse" is out of the scope of the
> > > draft. Otherwise the draft subject should more clear indicate what
> kind
> > > of reuse it refers to...
> > >
> > > Of course, once a single connection is established, according to the
>
> > > connect-reuse draft, it MAY be used for multiple sessions, but
> > > that's not the scope of the draft, so...
> > >
> > > Regards,
> > >
> > > Christer Holmberg
> > > Ericsson Finland
> >
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP Use
> > sip-implementors@cs.columbia.edu for questions on current sip Use
> > sip@ietf.org for new developments of core SIP


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From daemon@optimus.ietf.org  Thu Jul 11 11:09:27 2002
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Date: Thu, 11 Jul 2002 07:43:47 -0700
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        sipping@ietf.org
To: Christer Holmberg <christer.holmberg@lmf.ericsson.se>
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Subject: [Sip] Re: [Sipping] draft-mahy-sipping-connect-reuse-reqs-00
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Hi,

The current wording is that connection reuse is part of a request which 
can be rejected.  I need to understand Sean's use case a little better, 
but I suspect this will need to remain a request, because the request 
might not be authorized.  I guess the key question is what do you do  if 
the request can't be handled?  do you reject with an error message or 
handle the SIP request anyway (ignoring the connection reuse).  perhaps 
this is what Sean meant.  We probably should have an indication that the 
alias is optional or required.  Of course, required handling would also 
imply a Proxy-Require/Require which I was trying to avoid....

thanks,
-rohan

On Thursday, July 11, 2002, at 12:10 AM, Christer Holmberg wrote:

>
> Hi,
>
> I think it is important to think of who is allowed to "demand" what. 
> Can we
> always assume the client makes the final decission? What if the 
> server - for
> whatever reason - doens't want to/isn't able to reuse the connection? 
> Shall
> it send some kind of error response to the request where the "demand" is
> specified, or shall it be allowed to use another, or establish a new,
> connection? Shall the server be allowed to "demand" a re-use of 
> connection,
> even if the client didn't? How does this affect backward compability, 
> if the
> indicators aren't understood by all the nodes? Do we need
> Supported/Require/Proxy-Require tags?
>
> Regards,
>
> Christer Holmberg
> Ericsson Finland
>
>
>
>> Good point. Is this is a request or a demand by the client to re-use
>> the connection? I think we need the possibility for the client to
>> *demand* that the connection be re-used. There may also be situations
>> where re-use is just an optimization.
>>
>> /sean
>>
>> -----Original Message-----
>> From: Christer Holmberg [mailto:christer.holmberg@lmf.ericsson.se]
>> Sent: Tuesday, July 09, 2002 9:41 AM
>> To: Sean Olson; Gonzalo Camarillo; rohan@cisco.com; sip@ietf.org;
>> sipping@ietf.org
>> Subject: Re: [Sipping] draft-mahy-sipping-connect-reuse-reqs-00
>>
>> Another comment,
>>
>> I also think it is important to remember that there MAY be scenarios,
>> especially if a connection is used for multiple sessions, where one
>> wants to use separate connections per each direction - even if the
>> client indicates he wants to use the same connection for the response.
>> Just to think of before the MUSTs are added...
>>
>> Regards,
>>
>> Christer Holmberg
>> Ericsson Finland
>>
>> Christer Holmberg wrote:
>>
>>> Hi Sean,
>>>
>>>> Is this really out of scope for this draft? Naturally
>>>> the more re-use you make of the connection, the more benefit you get
>>
>>>> from re-use ;-)
>>>
>>> Of course, but in that case it should be mentioned in the draft :)
>>>
>>> The only thing the draft currently talks about the problem how to make
>>
>>> sure a single connection is used for both directions, per session -
>>> which is a separate issue from reusing connections for multiple
>>> sessions.
>>>
>>> And, since I don't think any draft currently talks about the reuse of
>>> connections for multiple sessions (bis talks about keeping the
>>> connection open to make sure the transaction uses the same
>>> transaction), maybe it could be added to the connect-reuse draft?
>>>
>>> Just some thoughts...
>>>
>>> Regards,
>>>
>>> Christer Holmberg
>>> Ericsson Finland
>>>
>>>>
>>>>
>>>> /sean
>>>>
>>>> -----Original Message-----
>>>> From: Christer Holmberg [mailto:christer.holmberg@lmf.ericsson.se]
>>>> Sent: Tuesday, July 09, 2002 6:30 AM
>>>> To: Gonzalo Camarillo
>>>> Cc: Sean Olson; rohan@cisco.com; sip@ietf.org; sipping@ietf.org
>>>> Subject: Re: [Sipping] draft-mahy-sipping-connect-reuse-reqs-00
>>>>
>>>> Hi,
>>>>
>>>> RFC3261/bis-9 mainly talks about using a single connection (or, one
>>>> in each
>>>> direction) for a single transaction, and the connect-reuse draft
>>>> talks about being able to use a single connection for messages in
>>>> both direction, for a single session. Sometimes, however, when we
>>>> talk about "reuse of connection" we mean when a single connection
>>>> (or, one in each
>>>> direction) is used for multiple sessions. I think it would be a good
>>>> idea to mention that that kind of "reuse" is out of the scope of the
>>>> draft. Otherwise the draft subject should more clear indicate what
>> kind
>>>> of reuse it refers to...
>>>>
>>>> Of course, once a single connection is established, according to the
>>
>>>> connect-reuse draft, it MAY be used for multiple sessions, but
>>>> that's not the scope of the draft, so...
>>>>
>>>> Regards,
>>>>
>>>> Christer Holmberg
>>>> Ericsson Finland
>>>
>>> _______________________________________________
>>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>>> This list is for NEW development of the application of SIP Use
>>> sip-implementors@cs.columbia.edu for questions on current sip Use
>>> sip@ietf.org for new developments of core SIP
>


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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Thread-Topic: [Sipping] draft-mahy-sipping-connect-reuse-reqs-00
Thread-Index: AcIo6UYXkv8CIiqhRDGMY/yr2C8UGAAEDLSQ
From: "Sean Olson" <seanol@windows.microsoft.com>
To: "Rohan Mahy" <rohan@cisco.com>,
        "Christer Holmberg" <christer.holmberg@lmf.ericsson.se>
Cc: "Gonzalo Camarillo" <Gonzalo.Camarillo@lmf.ericsson.se>, <sip@ietf.org>,
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Subject: [Sip] RE: [Sipping] draft-mahy-sipping-connect-reuse-reqs-00
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I am considering the use case where the connection between
the client and the server is "one-way". By one-way, I mean
only the client may establish a connection to the server and
not the other way around, usually due to an intermediate firewall.
In this case, it is critical that the server re-use the connection
that the client established to it, not only for this transaction
but for all subsequent messaging.

/sean

-----Original Message-----
From: Rohan Mahy [mailto:rohan@cisco.com] 
Sent: Thursday, July 11, 2002 7:44 AM
To: Christer Holmberg
Cc: Sean Olson; Gonzalo Camarillo; sip@ietf.org; sipping@ietf.org
Subject: Re: [Sipping] draft-mahy-sipping-connect-reuse-reqs-00


Hi,

The current wording is that connection reuse is part of a request which 
can be rejected.  I need to understand Sean's use case a little better, 
but I suspect this will need to remain a request, because the request 
might not be authorized.  I guess the key question is what do you do  if

the request can't be handled?  do you reject with an error message or 
handle the SIP request anyway (ignoring the connection reuse).  perhaps 
this is what Sean meant.  We probably should have an indication that the

alias is optional or required.  Of course, required handling would also 
imply a Proxy-Require/Require which I was trying to avoid....

thanks,
-rohan

On Thursday, July 11, 2002, at 12:10 AM, Christer Holmberg wrote:

>
> Hi,
>
> I think it is important to think of who is allowed to "demand" what.
> Can we
> always assume the client makes the final decission? What if the 
> server - for
> whatever reason - doens't want to/isn't able to reuse the connection? 
> Shall
> it send some kind of error response to the request where the "demand"
is
> specified, or shall it be allowed to use another, or establish a new,
> connection? Shall the server be allowed to "demand" a re-use of 
> connection,
> even if the client didn't? How does this affect backward compability, 
> if the
> indicators aren't understood by all the nodes? Do we need
> Supported/Require/Proxy-Require tags?
>
> Regards,
>
> Christer Holmberg
> Ericsson Finland
>
>
>
>> Good point. Is this is a request or a demand by the client to re-use 
>> the connection? I think we need the possibility for the client to
>> *demand* that the connection be re-used. There may also be situations

>> where re-use is just an optimization.
>>
>> /sean
>>
>> -----Original Message-----
>> From: Christer Holmberg [mailto:christer.holmberg@lmf.ericsson.se]
>> Sent: Tuesday, July 09, 2002 9:41 AM
>> To: Sean Olson; Gonzalo Camarillo; rohan@cisco.com; sip@ietf.org; 
>> sipping@ietf.org
>> Subject: Re: [Sipping] draft-mahy-sipping-connect-reuse-reqs-00
>>
>> Another comment,
>>
>> I also think it is important to remember that there MAY be scenarios,

>> especially if a connection is used for multiple sessions, where one 
>> wants to use separate connections per each direction - even if the 
>> client indicates he wants to use the same connection for the 
>> response. Just to think of before the MUSTs are added...
>>
>> Regards,
>>
>> Christer Holmberg
>> Ericsson Finland
>>
>> Christer Holmberg wrote:
>>
>>> Hi Sean,
>>>
>>>> Is this really out of scope for this draft? Naturally
>>>> the more re-use you make of the connection, the more benefit you 
>>>> get
>>
>>>> from re-use ;-)
>>>
>>> Of course, but in that case it should be mentioned in the draft :)
>>>
>>> The only thing the draft currently talks about the problem how to 
>>> make
>>
>>> sure a single connection is used for both directions, per session - 
>>> which is a separate issue from reusing connections for multiple 
>>> sessions.
>>>
>>> And, since I don't think any draft currently talks about the reuse 
>>> of connections for multiple sessions (bis talks about keeping the 
>>> connection open to make sure the transaction uses the same 
>>> transaction), maybe it could be added to the connect-reuse draft?
>>>
>>> Just some thoughts...
>>>
>>> Regards,
>>>
>>> Christer Holmberg
>>> Ericsson Finland
>>>
>>>>
>>>>
>>>> /sean
>>>>
>>>> -----Original Message-----
>>>> From: Christer Holmberg [mailto:christer.holmberg@lmf.ericsson.se]
>>>> Sent: Tuesday, July 09, 2002 6:30 AM
>>>> To: Gonzalo Camarillo
>>>> Cc: Sean Olson; rohan@cisco.com; sip@ietf.org; sipping@ietf.org
>>>> Subject: Re: [Sipping] draft-mahy-sipping-connect-reuse-reqs-00
>>>>
>>>> Hi,
>>>>
>>>> RFC3261/bis-9 mainly talks about using a single connection (or, one

>>>> in each
>>>> direction) for a single transaction, and the connect-reuse draft 
>>>> talks about being able to use a single connection for messages in 
>>>> both direction, for a single session. Sometimes, however, when we 
>>>> talk about "reuse of connection" we mean when a single connection 
>>>> (or, one in each
>>>> direction) is used for multiple sessions. I think it would be a 
>>>> good idea to mention that that kind of "reuse" is out of the scope 
>>>> of the draft. Otherwise the draft subject should more clear 
>>>> indicate what
>> kind
>>>> of reuse it refers to...
>>>>
>>>> Of course, once a single connection is established, according to 
>>>> the
>>
>>>> connect-reuse draft, it MAY be used for multiple sessions, but 
>>>> that's not the scope of the draft, so...
>>>>
>>>> Regards,
>>>>
>>>> Christer Holmberg
>>>> Ericsson Finland
>>>
>>> _______________________________________________
>>> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>>> This list is for NEW development of the application of SIP Use 
>>> sip-implementors@cs.columbia.edu for questions on current sip Use 
>>> sip@ietf.org for new developments of core SIP
>


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From daemon@optimus.ietf.org  Thu Jul 11 14:56:57 2002
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Date: Thu, 11 Jul 2002 11:15:58 -0700
From: Sarvi Shanmugham <sarvi@cisco.com>
Organization: Cisco Systems Inc.
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Is there a draft or spec addressing how out-of-band DTMF relay should be
done in SIP.

Is there preference between the INFO message and a NOTIFY message  to
carry this out-of-band DTMF between 2 SIP end-points.

Thanks,
Sarvi

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From daemon@optimus.ietf.org  Thu Jul 11 18:18:06 2002
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From: Kedar Patil <kpatil@sta.samsung.com>
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Hi,

Section 13.2.1 (Page 80) of RFC3261 says :

======================================================
o After having sent or received an answer to the first offer, the
  UAC MAY generate subsequent offers in requests based on rules
  specified for that method, but only "if it has received answers
  to any previous offers", AND "has not sent any offers to which it
  hasn't gotten an answer".
=======================================================

The last two clauses (enclosed in double quotes) joined by "AND"
mean the same thing, Isn't it? Is there any hidden case in either
clause not covered by the other?

thanks,
= Kedar =

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<P><FONT SIZE=2>Hi,</FONT>
</P>

<P><FONT SIZE=2>Section 13.2.1 (Page 80) of RFC3261 says :</FONT>
</P>

<P><FONT SIZE=2>======================================================</FONT>
<BR><FONT SIZE=2>o After having sent or received an answer to the first offer, the</FONT>
<BR><FONT SIZE=2>&nbsp; UAC MAY generate subsequent offers in requests based on rules</FONT>
<BR><FONT SIZE=2>&nbsp; specified for that method, but only &quot;if it has received answers</FONT>
<BR><FONT SIZE=2>&nbsp; to any previous offers&quot;, AND &quot;has not sent any offers to which it</FONT>
<BR><FONT SIZE=2>&nbsp; hasn't gotten an answer&quot;.</FONT>
<BR><FONT SIZE=2>=======================================================</FONT>
</P>

<P><FONT SIZE=2>The last two clauses (enclosed in double quotes) joined by &quot;AND&quot;</FONT>
<BR><FONT SIZE=2>mean the same thing, Isn't it? Is there any hidden case in either</FONT>
<BR><FONT SIZE=2>clause not covered by the other?</FONT>
</P>

<P><FONT SIZE=2>thanks,</FONT>
<BR><FONT SIZE=2>= Kedar =</FONT>
</P>

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Subject: [Sip] draft-ietf-sip-sec-agree-04.txt
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I have question regarding the purpose of the "Security-Client" field. Based 
on the description, it looks like this field is not used by the server in 
any way. Can someone explain the purpose of this field?

thanks,
AC

Ps: In the "Server Initiated" (Section 3.4.2) procedures, this field is not 
used. By the same reasoning, this field should not be applicable to the 
"Client Initiated" procedures as well.


At 12:33 PM 7/7/2002 -0400, Internet-Drafts@ietf.org wrote:
>A New Internet-Draft is available from the on-line Internet-Drafts 
>directories.
>This draft is a work item of the Session Initiation Protocol Working Group 
>of the IETF.
>
>         Title           : Security Mechanism Agreement for SIP Sessions
>         Author(s)       : J. Arkko et al.
>         Filename        : draft-ietf-sip-sec-agree-04.txt
>         Pages           : 17
>         Date            : 05-Jul-02
>
>SIP has a number of security mechanisms. Some of them have been built
>in to the SIP protocol, such as HTTP authentication or secure
>attachments. These mechanisms have even alternative algorithms and
>parameters. SIP does not currently provide any mechanism for
>selecting which security mechanisms to use between two entities. In
>particular, even if some mechanisms such as OPTIONS were used to make
>this selection, the selection would be vulnerable against the
>Bidding-Down attack. This document defines three header fields for
>negotiating the security mechanisms within SIP between a SIP entity
>and its next SIP hop. A SIP entity applying this mechanism must
>always require some minimum security (i.e. integrity protection) from
>all communicating parties in order to secure the negotiation, but the
>negotiation can agree on which specific minimum security is used.
>
>A URL for this Internet-Draft is:
>http://www.ietf.org/internet-drafts/draft-ietf-sip-sec-agree-04.txt
>
>To remove yourself from the IETF Announcement list, send a message to
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>
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>
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- Is defining a new sip-uri parameter considered to be an extension. This also applies to headers that allow extension params. If so, should this be mentioned in the draft?

- Should the draft also mention something about registering new sip headers, methods, option-tags with IANA?

- "Extensions that define new methods SHOULD use all capitals for the method name". Why not a MUST?

Regards,
Hisham

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From: "Sriram Parameswar" <sriramp@nortelnetworks.com>
To: "'Adam Roach'" <adam@dynamicsoft.com>,
        "'Rosen, Brian'"
	 <Brian.Rosen@marconi.com>,
        Robert Sparks <rsparks@dynamicsoft.com>
Cc: sip@ietf.org
Subject: RE: [Sip] REFER: Who has implemented REFER without creating subsc
	 riptions?
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Hi All,

I think there is definite value to having REFER without NOTIFYs. There are
several non transfer applications that benefit from REFER, without having
the overhead of the NOTIFY(s).

To this end, I suggest that we include an option-tag "nosubscription" to the
REFER method. Thus if a UA wants to send a REFER but does not care for
notification of success of failure - it puts in Require: nosubscription. The
inclusion of the 'nosubscription' option-tag causes the REFER to complete
upon receiving the 202, without the need for NOTIFY(s). This is also
backwards compatible - because a regular REFER (without this option-tag)
continues to show standard behavior with atleast one NOTIFY.

Comments are welcome,

Sriram
__________________________________________
Sriram Parameswar              Phone: 972-685-8540
Interactive Multimedia Server (IMS) Fax: 972-684-3986
Nortel Networks, Richardson USA  Email: sriramp@nortelnetworks.com


-----Original Message-----
From: Adam Roach [mailto:adam@dynamicsoft.com]
Sent: Wednesday, July 10, 2002 4:31 PM
To: 'Rosen, Brian'; Robert Sparks
Cc: sip@ietf.org
Subject: RE: [Sip] REFER: Who has implemented REFER without creating
subsc riptions?


> -----Original Message-----
> From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
>
> The new [REFER] spec requires at least two [NOTIFYs],
> an immediate one, and the one upon completion.

As Dan pointed out and Robert reiterated, this isn't
precisely true. You can get away with just one
NOTIFY: the immediate one. As long as the "Subscription-
State" header indicates "terminated," this behavior
complies with the draft. The distinction is important,
since it doesn't require recipients of REFER to implement
the full event mechanism.

/a

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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charset=3Diso-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2654.89">
<TITLE>RE: [Sip] REFER: Who has implemented REFER without creating =
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</HEAD>
<BODY>

<P><FONT SIZE=3D2>Hi All,</FONT>
</P>

<P><FONT SIZE=3D2>I think there is definite value to having REFER =
without NOTIFYs. There are several non transfer applications that =
benefit from REFER, without having the overhead of the =
NOTIFY(s).</FONT></P>

<P><FONT SIZE=3D2>To this end, I suggest that we include an option-tag =
&quot;nosubscription&quot; to the REFER method. Thus if a UA wants to =
send a REFER but does not care for notification of success of failure - =
it puts in Require: nosubscription. The inclusion of the =
'nosubscription' option-tag causes the REFER to complete upon receiving =
the 202, without the need for NOTIFY(s). This is also backwards =
compatible - because a regular REFER (without this option-tag) =
continues to show standard behavior with atleast one NOTIFY.</FONT></P>

<P><FONT SIZE=3D2>Comments are welcome,</FONT>
</P>

<P><FONT SIZE=3D2>Sriram</FONT>
<BR><FONT SIZE=3D2>__________________________________________</FONT>
<BR><FONT SIZE=3D2>Sriram =
Parameswar&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp; Phone: 972-685-8540</FONT>
<BR><FONT SIZE=3D2>Interactive Multimedia Server (IMS) Fax: =
972-684-3986</FONT>
<BR><FONT SIZE=3D2>Nortel Networks, Richardson USA&nbsp; Email: =
sriramp@nortelnetworks.com</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Adam Roach [<A =
HREF=3D"mailto:adam@dynamicsoft.com">mailto:adam@dynamicsoft.com</A>]</F=
ONT>
<BR><FONT SIZE=3D2>Sent: Wednesday, July 10, 2002 4:31 PM</FONT>
<BR><FONT SIZE=3D2>To: 'Rosen, Brian'; Robert Sparks</FONT>
<BR><FONT SIZE=3D2>Cc: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>Subject: RE: [Sip] REFER: Who has implemented REFER =
without creating</FONT>
<BR><FONT SIZE=3D2>subsc riptions?</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt; From: Rosen, Brian [<A =
HREF=3D"mailto:Brian.Rosen@marconi.com">mailto:Brian.Rosen@marconi.com</=
A>]</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt; The new [REFER] spec requires at least two =
[NOTIFYs],</FONT>
<BR><FONT SIZE=3D2>&gt; an immediate one, and the one upon =
completion.</FONT>
</P>

<P><FONT SIZE=3D2>As Dan pointed out and Robert reiterated, this =
isn't</FONT>
<BR><FONT SIZE=3D2>precisely true. You can get away with just =
one</FONT>
<BR><FONT SIZE=3D2>NOTIFY: the immediate one. As long as the =
&quot;Subscription-</FONT>
<BR><FONT SIZE=3D2>State&quot; header indicates &quot;terminated,&quot; =
this behavior</FONT>
<BR><FONT SIZE=3D2>complies with the draft. The distinction is =
important,</FONT>
<BR><FONT SIZE=3D2>since it doesn't require recipients of REFER to =
implement</FONT>
<BR><FONT SIZE=3D2>the full event mechanism.</FONT>
</P>

<P><FONT SIZE=3D2>/a</FONT>
</P>

<P><FONT =
SIZE=3D2>_______________________________________________</FONT>
<BR><FONT SIZE=3D2>Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>This list is for NEW development of the core SIP =
Protocol</FONT>
<BR><FONT SIZE=3D2>Use sip-implementors@cs.columbia.edu for questions =
on current sip</FONT>
<BR><FONT SIZE=3D2>Use sipping@ietf.org for new developments on the =
application of sip</FONT>
</P>

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_______________________________________________
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Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@ns.ietf.org  Sun Jul 14 08:19:55 2002
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From: "Ofir Arkin" <ofir@sys-security.com>
To: <sip@ietf.org>
Date: Sun, 14 Jul 2002 13:02:45 +0100
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Subject: [Sip] Some qustions on processing REGISTER requests
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While reading RFC 3261 section 10.3 Processing REGISTER requests I have
encountered several questions I could not find answers for:

1. Step 1: When the Registrar inspects the Request-URI to determine if
it has access to the required bindings, the RFC does not specify what
the Registrar should do if it does not have access to the binding (and
when it cannot forward the request). Is there a response code sent in
that case?

2. Step 5: the 'To' header value is examined (address-of-record) to
verify it is valid with the domain found in the Request-URI. The actual
verification test in not given with the text, but can I assume the
authors meant that the domain names should be the same?


Btw - Are there open source or free SIP Registrar servers that actually
implements the requirements found with RFC 3261?


Thanks!

Ofir Arkin [ofir@sys-security.com]
Founder
The Sys-Security Group
http://www.sys-security.com
PGP CC2C BE53 12C6 C9F2 87B1 B8C6 0DFA CF2D D360 43FA  




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From daemon@ns.ietf.org  Sun Jul 14 11:00:13 2002
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Date: Sun, 14 Jul 2002 23:39:27 +0900
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: Kedar Patil <kpatil@sta.samsung.com>
CC: "'sip@ietf.org'" <sip@ietf.org>
Subject: Re: [Sip] question on new offer in a dialog
References: <4B9386E83999D411997100508BAF206A061F4FAB@stamail.telecom.sna.samsung.com>
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Kedar Patil wrote:
> Hi, 
> 
> Section 13.2.1 (Page 80) of RFC3261 says : 
> 
> ====================================================== 
> o After having sent or received an answer to the first offer, the 
>   UAC MAY generate subsequent offers in requests based on rules 
>   specified for that method, but only "if it has received answers 
>   to any previous offers", AND "has not sent any offers to which it 
>   hasn't gotten an answer". 
> ======================================================= 
> 
> The last two clauses (enclosed in double quotes) joined by "AND" 
> mean the same thing, Isn't it? Is there any hidden case in either 
> clause not covered by the other? 

I think this is a bug in the spec. The first clause should read "if it 
has sent answers to any previous offers". Thanks for catching it.

-Jonathan R.



-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com


_______________________________________________
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From daemon@ns.ietf.org  Sun Jul 14 11:02:11 2002
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To: Sarvi Shanmugham <sarvi@cisco.com>
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Subject: Re: [Sip] Out-of-Band DTMF Relay
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Sarvi Shanmugham wrote:
> Is there a draft or spec addressing how out-of-band DTMF relay should be
> done in SIP.

This remains an active item of discussion. There are several drafts, 
which include:

http://search.ietf.org/internet-drafts/draft-rosenberg-sipping-markup-00.txt
http://search.ietf.org/internet-drafts/draft-fairlie-sipping-netapp-session-00.txt
http://search.ietf.org/internet-drafts/draft-culpepper-sip-key-events-01.txt

Unfortunately, there are many proprietary implementations for this as well.


> 
> Is there preference between the INFO message and a NOTIFY message  to
> carry this out-of-band DTMF between 2 SIP end-points.

INFO is really bad. I think there is general agreement on that. It 
causes extra signaling even if no one cares, and is limite in scope to 
just DTMF. All of the above three drafts use some mechanism whereby a 
server asks the client for input, which is generally better.

There is, of course, rfc2833, which is ideal for end-to-end DTMF, such 
as between gateways or between a PC and a gateway.

-Jonathan R.


-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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From daemon@ns.ietf.org  Sun Jul 14 11:07:22 2002
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hisham.khartabil@nokia.com wrote:
> - Is defining a new sip-uri parameter considered to be an extension.

Yes.

> This also applies to headers that allow extension params.

Yes.

> If so, should
> this be mentioned in the draft?

Yes. Its mentioned up front that these are extensions, but there is no 
real dsicussion of it. I will add it.


> 
> - Should the draft also mention something about registering new sip
> headers, methods, option-tags with IANA?

I think its worth mentioning, yes.

> 
> - "Extensions that define new methods SHOULD use all capitals for the
> method name". Why not a MUST?

Because, technically, they can be lowercase. Its just that we haven't 
had any, historically.

-Jonathan R.



-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com


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From daemon@ns.ietf.org  Sun Jul 14 11:32:01 2002
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Date: Mon, 15 Jul 2002 00:08:23 +0900
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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CC: Christer Holmberg <christer.holmberg@lmf.ericsson.se>,
        Gonzalo Camarillo
 <Gonzalo.Camarillo@lmf.ericsson.se>,
        rohan@cisco.com, sip@ietf.org, sipping@ietf.org
Subject: Re: [Sip] RE: [Sipping] draft-mahy-sipping-connect-reuse-reqs-00
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Sean Olson wrote:
> Good point. Is this is a request or a demand by the client to re-use
> the connection? I think we need the possibility for the client to
> *demand* that the connection be re-used. There may also be situations
> where re-use is just an optimization.

Yes, I agree.

My main reason for indicating that a connection MUST be reused is NATs. 
If I open a connection from my UA to the proxy through a NAT, I will 
only be able to receive incoming traffic on that connection. Therefore, 
reuse for incoming messages is a MUST, rather than just an optimiziation.

Another way to think about this is that the alias is "scoped". In the 
case of mandatory reuse, the alias is only defined on the element to 
which the request was sent. I suppose this means that its the 
responsibility of that element to make sure that requests addressed to 
that alias by some other entity get routed to that element first.

-Jonathan R.


> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@lmf.ericsson.se] 
> Sent: Tuesday, July 09, 2002 9:41 AM
> To: Sean Olson; Gonzalo Camarillo; rohan@cisco.com; sip@ietf.org;
> sipping@ietf.org
> Subject: Re: [Sipping] draft-mahy-sipping-connect-reuse-reqs-00
> 
> 
> 
> Another comment,
> 
> I also think it is important to remember that there MAY be scenarios,
> especially if a connection is used for multiple sessions, where one
> wants to use separate connections per each direction - even if the
> client indicates he wants to use the same connection for the response.
> Just to think of before the MUSTs are added...
> 
> Regards,
> 
> Christer Holmberg
> Ericsson Finland
> 
> 
> Christer Holmberg wrote:
> 
> 
>>Hi Sean,
>>
>>
>>>Is this really out of scope for this draft? Naturally
>>>the more re-use you make of the connection, the more benefit you get
>>
> 
>>>from re-use ;-)
>>
>>Of course, but in that case it should be mentioned in the draft :)
>>
>>The only thing the draft currently talks about the problem how to make
> 
> 
>>sure a single connection is used for both directions, per session - 
>>which is a separate issue from reusing connections for multiple 
>>sessions.
>>
>>And, since I don't think any draft currently talks about the reuse of 
>>connections for multiple sessions (bis talks about keeping the 
>>connection open to make sure the transaction uses the same 
>>transaction), maybe it could be added to the connect-reuse draft?
>>
>>Just some thoughts...
>>
>>Regards,
>>
>>Christer Holmberg
>>Ericsson Finland
>>
>>
>>>
>>>/sean
>>>
>>>-----Original Message-----
>>>From: Christer Holmberg [mailto:christer.holmberg@lmf.ericsson.se]
>>>Sent: Tuesday, July 09, 2002 6:30 AM
>>>To: Gonzalo Camarillo
>>>Cc: Sean Olson; rohan@cisco.com; sip@ietf.org; sipping@ietf.org
>>>Subject: Re: [Sipping] draft-mahy-sipping-connect-reuse-reqs-00
>>>
>>>Hi,
>>>
>>>RFC3261/bis-9 mainly talks about using a single connection (or, one 
>>>in each
>>>direction) for a single transaction, and the connect-reuse draft 
>>>talks about being able to use a single connection for messages in 
>>>both direction, for a single session. Sometimes, however, when we 
>>>talk about "reuse of connection" we mean when a single connection 
>>>(or, one in each
>>>direction) is used for multiple sessions. I think it would be a good
>>>idea to mention that that kind of "reuse" is out of the scope of the
>>>draft. Otherwise the draft subject should more clear indicate what
>>
> kind
> 
>>>of reuse it refers to...
>>>
>>>Of course, once a single connection is established, according to the
>>
> 
>>>connect-reuse draft, it MAY be used for multiple sessions, but 
>>>that's not the scope of the draft, so...
>>>
>>>Regards,
>>>
>>>Christer Holmberg
>>>Ericsson Finland
>>
>>_______________________________________________
>>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>>This list is for NEW development of the application of SIP Use 
>>sip-implementors@cs.columbia.edu for questions on current sip Use 
>>sip@ietf.org for new developments of core SIP
> 
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 


-- 
Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
Chief Scientist                         First Floor
dynamicsoft                             East Hanover, NJ 07936
jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
http://www.jdrosen.net                  PH:  (973) 952-5000
http://www.dynamicsoft.com


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Sun Jul 14 19:12:34 2002
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        Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>, rohan@cisco.com,
        sip@ietf.org, sipping@ietf.org
Subject: Re: [Sip] RE: [Sipping] draft-mahy-sipping-connect-reuse-reqs-00
References: <F66A04C29AD9034A8205949AD0C9010403FCEBC2@win-msg-02.wingroup.windeploy.ntdev.microsoft.com> <3D3193E7.2070607@dynamicsoft.com>
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Hi,

We should also look from the other direction, and remember that node
establishing the connection may also be a proxy forwarding an INVITE to a SIP
UA, requiring the connection to be reused in both direction for the purpose
of "saving" connections. Or, it may of course be a NAT. So, from that point
of view it is good that the client can make the reuse "decission".

However, I also think it is good if the called party, receiving the INVITE,
can require the connection to be used. For example, a single-session SIP UA
making a call may not care if it has to establish one or two connections. It
may, however, send the INVITE to a proxy, and/or a NAT, which again does
require the connection to be reused, so it should be able to do so.

I also think it is important to have a look how the solution will affect the
reuse of connections for multiple sessions (IF eg some extra parameters are
needed for that, or if some clarifications are needed).

Regards,

Christer Holmberg
Ericsson Finland



Jonathan Rosenberg wrote:

> Sean Olson wrote:
> > Good point. Is this is a request or a demand by the client to re-use
> > the connection? I think we need the possibility for the client to
> > *demand* that the connection be re-used. There may also be situations
> > where re-use is just an optimization.
>
> Yes, I agree.
>
> My main reason for indicating that a connection MUST be reused is NATs.
> If I open a connection from my UA to the proxy through a NAT, I will
> only be able to receive incoming traffic on that connection. Therefore,
> reuse for incoming messages is a MUST, rather than just an optimiziation.
>
> Another way to think about this is that the alias is "scoped". In the
> case of mandatory reuse, the alias is only defined on the element to
> which the request was sent. I suppose this means that its the
> responsibility of that element to make sure that requests addressed to
> that alias by some other entity get routed to that element first.
>
> -Jonathan R.
>
> > -----Original Message-----
> > From: Christer Holmberg [mailto:christer.holmberg@lmf.ericsson.se]
> > Sent: Tuesday, July 09, 2002 9:41 AM
> > To: Sean Olson; Gonzalo Camarillo; rohan@cisco.com; sip@ietf.org;
> > sipping@ietf.org
> > Subject: Re: [Sipping] draft-mahy-sipping-connect-reuse-reqs-00
> >
> >
> >
> > Another comment,
> >
> > I also think it is important to remember that there MAY be scenarios,
> > especially if a connection is used for multiple sessions, where one
> > wants to use separate connections per each direction - even if the
> > client indicates he wants to use the same connection for the response.
> > Just to think of before the MUSTs are added...
> >
> > Regards,
> >
> > Christer Holmberg
> > Ericsson Finland
> >
> >
> > Christer Holmberg wrote:
> >
> >
> >>Hi Sean,
> >>
> >>
> >>>Is this really out of scope for this draft? Naturally
> >>>the more re-use you make of the connection, the more benefit you get
> >>
> >
> >>>from re-use ;-)
> >>
> >>Of course, but in that case it should be mentioned in the draft :)
> >>
> >>The only thing the draft currently talks about the problem how to make
> >
> >
> >>sure a single connection is used for both directions, per session -
> >>which is a separate issue from reusing connections for multiple
> >>sessions.
> >>
> >>And, since I don't think any draft currently talks about the reuse of
> >>connections for multiple sessions (bis talks about keeping the
> >>connection open to make sure the transaction uses the same
> >>transaction), maybe it could be added to the connect-reuse draft?
> >>
> >>Just some thoughts...
> >>
> >>Regards,
> >>
> >>Christer Holmberg
> >>Ericsson Finland
> >>
> >>
> >>>
> >>>/sean
> >>>
> >>>-----Original Message-----
> >>>From: Christer Holmberg [mailto:christer.holmberg@lmf.ericsson.se]
> >>>Sent: Tuesday, July 09, 2002 6:30 AM
> >>>To: Gonzalo Camarillo
> >>>Cc: Sean Olson; rohan@cisco.com; sip@ietf.org; sipping@ietf.org
> >>>Subject: Re: [Sipping] draft-mahy-sipping-connect-reuse-reqs-00
> >>>
> >>>Hi,
> >>>
> >>>RFC3261/bis-9 mainly talks about using a single connection (or, one
> >>>in each
> >>>direction) for a single transaction, and the connect-reuse draft
> >>>talks about being able to use a single connection for messages in
> >>>both direction, for a single session. Sometimes, however, when we
> >>>talk about "reuse of connection" we mean when a single connection
> >>>(or, one in each
> >>>direction) is used for multiple sessions. I think it would be a good
> >>>idea to mention that that kind of "reuse" is out of the scope of the
> >>>draft. Otherwise the draft subject should more clear indicate what
> >>
> > kind
> >
> >>>of reuse it refers to...
> >>>
> >>>Of course, once a single connection is established, according to the
> >>
> >
> >>>connect-reuse draft, it MAY be used for multiple sessions, but
> >>>that's not the scope of the draft, so...
> >>>
> >>>Regards,
> >>>
> >>>Christer Holmberg
> >>>Ericsson Finland
> >>
> >>_______________________________________________
> >>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> >>This list is for NEW development of the application of SIP Use
> >>sip-implementors@cs.columbia.edu for questions on current sip Use
> >>sip@ietf.org for new developments of core SIP
> >
> >
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
>
> --
> Jonathan D. Rosenberg, Ph.D.            72 Eagle Rock Avenue
> Chief Scientist                         First Floor
> dynamicsoft                             East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                 FAX: (973) 952-5050
> http://www.jdrosen.net                  PH:  (973) 952-5000
> http://www.dynamicsoft.com


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Sun Jul 14 21:39:34 2002
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From: Adam Roach <adam@dynamicsoft.com>
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        Jonathan Rosenberg
	 <jdrosen@dynamicsoft.com>,
        Cullen Jennings <fluffy@cisco.com>
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Subject: RE: [Sip] Proposed change in draft-ietf-sip-nat
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> -----Original Message-----
> From: Pete Cordell [mailto:pcordell@ridgewaysystems.com]
>
> So why not acknowledge the editorial error (which
> requires no action) and move on.

Wow. I never thought I'd see the day that the SIP Working Group
mailing list devolved to the level of an old-fashioned Usenet-style
spelling flame.

Thanks, Pete. It's a first.

/a

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This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip



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From: Adam Roach <adam@dynamicsoft.com>
To: jh@lohi.eng.song.fi, "Peterson, Jon" <jon.peterson@neustar.biz>
Cc: Rohan Mahy <rohan@cisco.com>, sip@ietf.org
Subject: RE: [Sip] P- headers vs. requirements in sip change process
Date: Sun, 14 Jul 2002 20:15:12 -0500
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Catching up on old mail...

> -----Original Message-----
> From: jh@lohi.eng.song.fi [mailto:jh@lohi.eng.song.fi]
>
> also, nai is and in fact MUST be used also within a single 
> trust domain
> if the user has specified anonymous from header and the 
> request goes to
> a sip/pstn gw..  so the nai header is not just some "nice to know"
> thing, but an integral part of the sip specification.

Why? The user's identity can be ascertained from the credentials
(the username, in particular) provided at authentication time.

/a

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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From daemon@optimus.ietf.org  Sun Jul 14 21:43:21 2002
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From: Adam Roach <adam@dynamicsoft.com>
To: hisham.khartabil@nokia.com, sip@ietf.org
Subject: RE: [Sip] Questions on sip-guidelines
Date: Sun, 14 Jul 2002 20:15:06 -0500
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> -----Original Message-----
> From: hisham.khartabil@nokia.com [mailto:hisham.khartabil@nokia.com]
> 
> - Is defining a new sip-uri parameter considered to be an 
> extension. This also applies to headers that allow extension 
> params.

Yes, these are extensions. In particular, just adding on your own
made-up, private URI parameters runs the chances of colliding
with someone else's made-up, private URI parameters (which almost
certainly have different semantics than yours). So, you can't do
that and still claim to be SIP compliant. You'll need to at least
register the new parameters with IANA to avoid such collisions, and
the tsvchange document would seem to indicate that doing so requires
a SIP working group document.

/a

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From daemon@optimus.ietf.org  Sun Jul 14 21:48:06 2002
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From: Adam Roach <adam@dynamicsoft.com>
To: Sriram Parameswar <sriramp@nortelnetworks.com>,
        "'Adam Roach'"
	 <adam@dynamicsoft.com>,
        "'Rosen, Brian'" <Brian.Rosen@marconi.com>,
        Robert Sparks <rsparks@dynamicsoft.com>
Cc: sip@ietf.org
Subject: RE: [Sip] REFER: Who has implemented REFER without creating subsc
	 riptions?
Date: Sun, 14 Jul 2002 20:15:07 -0500
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>-----Original Message-----
>From: Sriram Parameswar [mailto:sriramp@nortelnetworks.com]
>
>I think there is definite value to having REFER without NOTIFYs.
>There are several non transfer applications that benefit from
>REFER, without having the overhead of the NOTIFY(s).

In particular, this reasoning flies in the face of the principle
of selecting "generality over efficiency" that has been established
for the SIP protocol and its extensions.

/a

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Use sip-implementors@cs.columbia.edu for questions on current sip
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From daemon@optimus.ietf.org  Mon Jul 15 00:42:38 2002
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To: Adam Roach <adam@dynamicsoft.com>
Cc: "Peterson, Jon" <jon.peterson@neustar.biz>, Rohan Mahy <rohan@cisco.com>,
        sip@ietf.org
Subject: RE: [Sip] P- headers vs. requirements in sip change process
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Adam Roach writes:

 > > also, nai is and in fact MUST be used also within a single 
 > > trust domain
 > > if the user has specified anonymous from header and the 
 > > request goes to
 > > a sip/pstn gw..  so the nai header is not just some "nice to know"
 > > thing, but an integral part of the sip specification.
 > 
 > Why? The user's identity can be ascertained from the credentials
 > (the username, in particular) provided at authentication time.

because one user may have many sip identities.  this should have been
clear based on several earlier emails about this topic.

-- juha

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From daemon@optimus.ietf.org  Mon Jul 15 01:34:02 2002
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From: "Sriram Parameswar" <sriramp@nortelnetworks.com>
To: "'Adam Roach'" <adam@dynamicsoft.com>,
        "'Rosen, Brian'"
	 <Brian.Rosen@marconi.com>,
        Robert Sparks <rsparks@dynamicsoft.com>
Cc: sip@ietf.org
Subject: RE: [Sip] REFER: Who has implemented REFER without creating subsc
	 riptions?
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I am afraid I do not agree with the rather general statement you have made
below. Implicit subscription created by REFER for all REFERs is does not
rate as a generality. It is a special use of Implicit Subscriptions. There
are applications that could 'refer' the recipient to a resource, but not
care to receive a notification. 

Thx,

Sriram 


-----Original Message-----
From: Adam Roach [mailto:adam@dynamicsoft.com]
Sent: Sunday, July 14, 2002 8:15 PM
To: Parameswar, Sriram [NGC:B615:EXCH]; 'Adam Roach'; 'Rosen, Brian';
Robert Sparks
Cc: sip@ietf.org
Subject: RE: [Sip] REFER: Who has implemented REFER without creating
subsc riptions?


>-----Original Message-----
>From: Sriram Parameswar [mailto:sriramp@nortelnetworks.com]
>
>I think there is definite value to having REFER without NOTIFYs.
>There are several non transfer applications that benefit from
>REFER, without having the overhead of the NOTIFY(s).

In particular, this reasoning flies in the face of the principle
of selecting "generality over efficiency" that has been established
for the SIP protocol and its extensions.

/a

------_=_NextPart_001_01C22BBE.6E6A0E20
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	charset="iso-8859-1"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2654.89">
<TITLE>RE: [Sip] REFER: Who has implemented REFER without creating =
subsc riptions?</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>I am afraid I do not agree with the rather general =
statement you have made below. Implicit subscription created by REFER =
for all REFERs is does not rate as a generality. It is a special use of =
Implicit Subscriptions. There are applications that could 'refer' the =
recipient to a resource, but not care to receive a notification. =
</FONT></P>

<P><FONT SIZE=3D2>Thx,</FONT>
</P>

<P><FONT SIZE=3D2>Sriram </FONT>
</P>
<BR>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Adam Roach [<A =
HREF=3D"mailto:adam@dynamicsoft.com">mailto:adam@dynamicsoft.com</A>]</F=
ONT>
<BR><FONT SIZE=3D2>Sent: Sunday, July 14, 2002 8:15 PM</FONT>
<BR><FONT SIZE=3D2>To: Parameswar, Sriram [NGC:B615:EXCH]; 'Adam =
Roach'; 'Rosen, Brian';</FONT>
<BR><FONT SIZE=3D2>Robert Sparks</FONT>
<BR><FONT SIZE=3D2>Cc: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>Subject: RE: [Sip] REFER: Who has implemented REFER =
without creating</FONT>
<BR><FONT SIZE=3D2>subsc riptions?</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>&gt;-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>&gt;From: Sriram Parameswar [<A =
HREF=3D"mailto:sriramp@nortelnetworks.com">mailto:sriramp@nortelnetworks=
.com</A>]</FONT>
<BR><FONT SIZE=3D2>&gt;</FONT>
<BR><FONT SIZE=3D2>&gt;I think there is definite value to having REFER =
without NOTIFYs.</FONT>
<BR><FONT SIZE=3D2>&gt;There are several non transfer applications that =
benefit from</FONT>
<BR><FONT SIZE=3D2>&gt;REFER, without having the overhead of the =
NOTIFY(s).</FONT>
</P>

<P><FONT SIZE=3D2>In particular, this reasoning flies in the face of =
the principle</FONT>
<BR><FONT SIZE=3D2>of selecting &quot;generality over efficiency&quot; =
that has been established</FONT>
<BR><FONT SIZE=3D2>for the SIP protocol and its extensions.</FONT>
</P>

<P><FONT SIZE=3D2>/a</FONT>
</P>

</BODY>
</HTML>
------_=_NextPart_001_01C22BBE.6E6A0E20--

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From daemon@optimus.ietf.org  Mon Jul 15 02:02:18 2002
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Subject: RE: [Sip] REFER: Who has implemented REFER without creating subsc
	riptions?
From: Robert Sparks <rsparks@dynamicsoft.com>
To: Sriram Parameswar <sriramp@nortelnetworks.com>
Cc: "'Adam Roach'" <adam@dynamicsoft.com>,
        "'Rosen, Brian'"
	 <Brian.Rosen@marconi.com>, sip@ietf.org
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	<EF1056F8EB4ED511B8FB0002A56079D401E5B126@zrc2c014.us.nortel.com>
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	<EF1056F8EB4ED511B8FB0002A56079D401E5B126@zrc2c014.us.nortel.com>
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Going with a mandatory creation of a subscription for the base
REFER document really simplifies the spec and implementation of it.

If you have an application that truly cannot stomach getting exactly
one notify, I suggest we work out an extension later.

Note that as a REFERor, if you don't care about the notifies, all
you have to do is reject the first one. That terminates the
subscription.

If you are a REFERee and you don't want to deal with holding onto
subscription state, send exactly one NOTIFY with a terminated
subscription-state.

RjS

On Mon, 2002-07-15 at 14:14, Sriram Parameswar wrote:
> I am afraid I do not agree with the rather general statement you have
> made below. Implicit subscription created by REFER for all REFERs is
> does not rate as a generality. It is a special use of Implicit
> Subscriptions. There are applications that could 'refer' the recipient
> to a resource, but not care to receive a notification. 
> 
> Thx, 
> 
> Sriram 
> 
> 
> -----Original Message----- 
> From: Adam Roach [ mailto:adam@dynamicsoft.com
> <mailto:adam@dynamicsoft.com> ] 
> Sent: Sunday, July 14, 2002 8:15 PM 
> To: Parameswar, Sriram [NGC:B615:EXCH]; 'Adam Roach'; 'Rosen, Brian'; 
> Robert Sparks 
> Cc: sip@ietf.org 
> Subject: RE: [Sip] REFER: Who has implemented REFER without creating 
> subsc riptions? 
> 
> 
> >-----Original Message----- 
> >From: Sriram Parameswar [ mailto:sriramp@nortelnetworks.com
> <mailto:sriramp@nortelnetworks.com> ] 
> > 
> >I think there is definite value to having REFER without NOTIFYs. 
> >There are several non transfer applications that benefit from 
> >REFER, without having the overhead of the NOTIFY(s). 
> 
> In particular, this reasoning flies in the face of the principle 
> of selecting "generality over efficiency" that has been established 
> for the SIP protocol and its extensions. 
> 
> /a 
> 



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-----Original Message-----
From: Christer Holmberg [mailto:christer.holmberg@lmf.ericsson.se] 
Sent: Monday, July 01, 2002 8:15 PM
To: Mark Watson
Cc: Alexeitsev, D; sip@ietf.org; Chris Hogg
Subject: Re: [Sip] Local vs Remote ringback on 180



Hi Mark,

> I'm confused by this discussion, because there seems to be a desire to 
> extrapolate some semantics about the *media* from a message which does 
> not contain SDP. Apologies if I have missed the issue at hand, or if 
> the following is obvious, but...
>
> The unify model means that media and session state are completely 
> independent, so, to talk about '180 with SDP' or '180 without SDP' 
> makes no sense.

I agree, but those are the terms used by the drafts I've refered to. The ISUP/SIP mapping draft says that "180 with SDP" means that a remote ringback will be generated, and bis-09 says that "180 without SDP" MAY cause the UAC to generate a local ringback.

Of course, those statements may sometimes be valid, if we receive either a 180 without SDP, OR we receive 180 with SDP - and no early dialog still exists. My orignial question was what we do if we receive 180 without SDP, but we DO have an early dialog established. It has been argued that the UAC should not generate local ringback in that case - instead assume the UAS does it since there is an early dialog established (if the UAS wants the UAC to generate local ringback it would have to put the media on hold using UPDATE). IF that's the case, I think it should me more clarified. Clarification is the only thing I've asked for - no changes in the protocol.

> You can talk about the effect 180 has on the session state (indicates 
> that called party is being alerted), but this has nothing to do with 
> the media state (if there was media before, there still is, if there 
> was no media before, there still isn't).

I've said all the time that the session state is not affected, only that the 180 affects what the application does with the media.

So, what you just wrote means, that in the case we have an established early dialog it really doesn't matter, from a ringback point of view, if there is a SDP in the 180 or not. Is that the "official" clarification I've asked for? :)

> You can talk about the effect that SDP offer/answer exchanges have on 
> the media, but this operates independently of the SIP message those 
> SDP bodies arrive in.

True. That is why I also asked if the SDP is a good "ringback indicator" in the first place...

> So,
>
> 1) the UAC knows whether there is media or not (based on SDP 
> offer/answer exchanges so far)
> 2) if 180 has been received, the UAC knows that the called party is 
> alerting
>
> If the UAC knows that the called party is alerting AND knows that 
> there is media, it is a local matter what to do. Personally, I cannot 
> see an argument for doing anything other than playing the media (it 
> might contain 'ringing with call waiting announcement').

I think it was earlier said that the decission can't be based on if there is media or not. There may be RTP packets, but that doesn't mean they contain any "real" media.

> A PROBLEM arises if there is a UAS which negotiates media but then 
> sends silence (no packets) - there ARE some reasons why it might want 
> to do this to do with reducing clipping on answer. The result is that 
> silence would be all the caller would hear even though the UAC may 
> know that the called party is being alerted. I think we just have to 
> say that it is foolish for the UAS to negotiate media and then send 
> silence - it should put the media on hold if it has nothing to send.

Like I said earlier, that would probably not happend when the same server establishes the early dialog and sends the 180.  I earlier gave an example where a B2BUA sends the announcement ("you have now reached the X company"), the contacts a SIP phone which sends 180 (the SIP phone does NOT generate early media). Now, the B2BUA gets the 180, stop playing the announcement, and forwards the 180 to the UAC. Now, since there are no more announcements, and the SIP phone doesn't generate remote ringback, and the UAC doesn't generate local ringback, there is no ringback at all. In this case the B2BUA would have to send UPDATE (media on hold), then forward the 180 (the UAC will now generate local ringback), and when the SIP phone answers it would have to send another UPDATE (media un-hold) and then forward the 200.

An issue related to this I brought up earlier, is if the UAS will have to buffer the 200 until both UPDATE transactions have been completed. Any comments on that?

Also, in a non-PRACK environment, is a 18x "allowed" to establish an early dialog in the first place? Bis-09 says that the answer must be sent in a reliable response, which is 200 OK. Yes, bis-09 DOES say that the answer MAY also be sent earlier, unreliably in a 18x, but when is the early dialog actually "officially" established? Again, the UAC could make a ringback decission based on if there is a SDP in the 180 or not, but a unreliably sent 180 may not even arrive...


Regards,

Christer Holmberg
Ericsson Finland



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From daemon@optimus.ietf.org  Mon Jul 15 05:40:05 2002
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Subject: [Sip] The cost of UDP
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Was caught in a middle of a row during the meeting and could not get to 
the microphone.

Wanted to remark on the remark that UDP is less resource consuming then 
TCP and you can
get more users on a machine with UDP.

Well, it is very much dependent on your implementation. I have seen an 
implementation that can
hold 20-30K users on a simple NT machine quite easily. Saying that UDP 
enable you to support
more users is a bit misleading what happens is that you consume less 
system resources but you
end up in consuming much more application resources since now you need to 
do what TCP does
for you in the application level. Using TCP and TLS also goes with the 
lines of using existing
protocols to solve problems and not having to solve these problems again 
and again in the
applications.

As can be understood, I am all for deprecating UDP while SIP is still 
young and it can be done.
Regarding backward compatibility issues, I am not so worried about it. I 
am not aware of
a very large SIP deployments that are used by millions of users currently. 
It will be relatively
easy to modify proxies and servers to accept and initiate only TCP. 
Regarding clients, in the time frame that we
are speaking about, 12-18 months, the clients can also be modified.

Avshalom


--=_alternative 00323DDEC2256BF7_=
Content-Type: text/html; charset="us-ascii"


<br><font size=2 face="sans-serif">Was caught in a middle of a row during the meeting and could not get to the microphone.</font>
<br>
<br><font size=2 face="sans-serif">Wanted to remark on the remark that UDP is less resource consuming then TCP and you can</font>
<br><font size=2 face="sans-serif">get more users on a machine with UDP.</font>
<br>
<br><font size=2 face="sans-serif">Well, it is very much dependent on your implementation. I have seen an implementation that can</font>
<br><font size=2 face="sans-serif">hold 20-30K users on a simple NT machine quite easily. Saying that UDP enable you to support</font>
<br><font size=2 face="sans-serif">more users is a bit misleading what happens is that you consume less system resources but you</font>
<br><font size=2 face="sans-serif">end up in consuming much more application resources since now you need to do what TCP does</font>
<br><font size=2 face="sans-serif">for you in the application level. Using TCP and TLS also goes with the lines of using existing</font>
<br><font size=2 face="sans-serif">protocols to solve problems and not having to solve these problems again and again in the</font>
<br><font size=2 face="sans-serif">applications.</font>
<br>
<br><font size=2 face="sans-serif">As can be understood, I am all for deprecating UDP while SIP is still young and it can be done.</font>
<br><font size=2 face="sans-serif">Regarding backward compatibility issues, I am not so worried about it. I am not aware of</font>
<br><font size=2 face="sans-serif">a very large SIP deployments that are used by millions of users currently. It will be relatively</font>
<br><font size=2 face="sans-serif">easy to modify proxies and servers to accept and initiate only TCP. Regarding clients, in the time frame that we</font>
<br><font size=2 face="sans-serif">are speaking about, 12-18 months, the clients can also be modified.</font>
<br>
<br><font size=2 face="sans-serif">Avshalom</font>
<br><font size=2 face="sans-serif"><br>
</font>
--=_alternative 00323DDEC2256BF7_=--

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From daemon@optimus.ietf.org  Mon Jul 15 11:56:30 2002
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From: "Valerie Gastaud" <valerie.gastaud@ulticom.com>
To: "Sip" <sip@ietf.org>
Date: Mon, 15 Jul 2002 17:27:00 +0200
Message-ID: <NDBBIHAOCKGAMIKNLALOCELFDGAA.valerie.gastaud@ulticom.com>
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Subject: [Sip] SIP draft-bis-09 to RFC 3261
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Hi,
Can someone tell me what are the (major) changes between 2543-bis09 and RFC
3261?
Thanks,
Valerie



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From daemon@optimus.ietf.org  Mon Jul 15 16:10:27 2002
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To: "Valerie Gastaud" <valerie.gastaud@ulticom.com>, "Sip" <sip@ietf.org>
From: Jiri Kuthan <kuthan@fokus.gmd.de>
Subject: Re: [Sip] SIP draft-bis-09 to RFC 3261
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You may want to look at wdiff at iptel.org archive.
http://www.iptel.org/info/players/ietf/callsignalling/draft-ietf-sip-rfc2543bis-09.txt.wdiff

-Jiri

At 05:27 PM 7/15/2002, Valerie Gastaud wrote:
>Hi,
>Can someone tell me what are the (major) changes between 2543-bis09 and RFC
>3261?
>Thanks,
>Valerie
>
>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip 


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From daemon@optimus.ietf.org  Tue Jul 16 01:01:25 2002
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Subject: [Sip] query on conference model
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Hi
      according to draft draft-ietf-sipping-service-examples-02.txt , user B
sends INVITE to bridge to establish session. Now how do I differentiate this
INVITE from the normal INVITE?
I have this problem because my conference server is a combination of normal
UAS, Registrar and conference server.

Thanks In Advance

Ranjit


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From: ABDUL SATTAR SHAIK <abdulss@migv.mot.com>
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Hi
what are the differences between rfc2543-bis-09 and rfc 3261?

I  didn't find a single difference in these two.

I think the bis-09 is  published as 3261 since bis-09 expires by august.

AmI right ?

Thanks
abdul sattar



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<P><FONT SIZE=2 FACE="Arial">Hi</FONT>
<BR><FONT SIZE=2 FACE="Arial">what are the differences between rfc2543-bis-09 and rfc 3261?</FONT>
</P>

<P><FONT SIZE=2 FACE="Arial">I&nbsp; didn't find a single difference in these two.</FONT>
</P>

<P><FONT SIZE=2 FACE="Arial">I think the bis-09 is&nbsp; published as 3261 since bis-09 expires by august.</FONT>
</P>

<P><FONT SIZE=2 FACE="Arial">AmI right ?</FONT>
</P>

<P><FONT SIZE=2 FACE="Arial">Thanks</FONT>
<BR><FONT SIZE=2 FACE="Arial">abdul sattar</FONT>
</P>
<BR>

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From daemon@optimus.ietf.org  Tue Jul 16 03:32:15 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Avshalom Houri'" <AVSHALOM@il.ibm.com>, <sip@ietf.org>
Subject: RE: [Sip] The cost of UDP
Date: Tue, 16 Jul 2002 02:00:36 -0500
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Avshalom said:
--------------
Wanted to remark on the remark that UDP is less resource consuming then
TCP and you can 
get more users on a machine with UDP. 

Well, it is very much dependent on your implementation. I have seen an
implementation that can 
hold 20-30K users on a simple NT machine quite easily. Saying that UDP
enable you to support 
more users is a bit misleading what happens is that you consume less
system resources but you 
end up in consuming much more application resources since now you need
to do what TCP does 
for you in the application level. Using TCP and TLS also goes with the
lines of using existing 
protocols to solve problems and not having to solve these problems again
and again in the 
applications. 
---------------

The "TCP limit" is based on the fact that most sockets-based
implementations of TCP use up one filesystem file handle per connected
socket. Since most kernel limitations use an unsigned 16-bit index for
the filehandle table, this puts a fairly hard upper bound of 64k
connected sockets. Given that one is likely to need a few to support the
OS and other applications, this makes a firm upper bound of around
64,000. Of course, many operating systems don't actually support this
many, with many TCP implementations actually breakikng at a much smaller
number, perhaps 16 or 256, but we're not likely to see commercial-grade
proxy systems built on these systems.


Of course, with connection-reuse in a proxy-proxy scenario, one socket
might actually be supporting thousands of users. The real issue is on
"boundary" proxies, where there is fan-in (or fan-out) to many UAs or
small (residential?) proxies.

What's the impact?

Let's say we average one 7-message dialog sequence per hour per device.
That's 7/messages per hour, or about 2 messages per second per 1,000 UA.
With an average proxy capable of handling 1,000 messages per second, we
should be able to handle 500,000 UA per proxy. That's clearly a larger
number than 64k, so there IS a potential for fan in/out limitations
being induced by TCP on larger servers.  Of course, if one factors in
presence and a little more traffic, it's not at all unreasoablle to
expect more than 70 messages per hour per UA, which with this size proxy
gets us CPU bound at around 50,000 clients, so the TCP constraint isn't
binding.

In other words, my analysis seems to indicate that for any "reasonable"
model of proxy fanout that the TCP connected socket limitation is only
to likely to impact really BIG proxy servers using a monolithic
"mainframe" style architecture not common in today's distributed
environments. In other words, it's just not much of a problem.

Another aspect to consider is the "half open connection limit". There
exists a constraint in many TCP implementations as to how many sockets
may be in a connecting (not connected) state at any given time. Just a
few years back, this number was something like 16 in a certain "robust
commercial UNIX". I think it's commonly much larger now. This was highly
problematic with HTTP 1.0 and it's very short-lived TCP connections, but
our model of persistent TCP/TLS connections and connection reuse greatly
reduce this liability.

Consider that 50,000 client server. If every single TCP connection is
broken and restored over the course of an hour, that gives us something
like 5 half/open connections at any given time. If however every
connection is restarted at exactly the same time, we DO have a bit more
of a problem, so there may well be an issue in this regard.

--
Dean


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From daemon@optimus.ietf.org  Tue Jul 16 04:04:11 2002
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'ABDUL SATTAR SHAIK'" <abdulss@migv.mot.com>,
        "'sip@ietf.org'"
	 <sip@ietf.org>
Subject: RE: [Sip] (no subject)
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There were a number of relatively small things fixed. 
For example, several ABNF errors were reported after bis-09
and before RFC3261 which were fixed.   I believe we will
provide a diff to highlight the changes.  This is harder than
you might think because the RFC editor completely redid the
formatting, so creating a diff is not really easy.  
 
There are no actual protocol changes.
 
Brian

-----Original Message-----
From: ABDUL SATTAR SHAIK [mailto:abdulss@migv.mot.com]
Sent: Tuesday, July 16, 2002 2:06 AM
To: 'sip@ietf.org'
Subject: [Sip] (no subject)



Hi 
what are the differences between rfc2543-bis-09 and rfc 3261? 

I  didn't find a single difference in these two. 

I think the bis-09 is  published as 3261 since bis-09 expires by august. 

AmI right ? 

Thanks 
abdul sattar 



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<DIV><SPAN class=857403607-16072002><FONT face=Arial color=#0000ff>There were a 
number of relatively small things fixed.&nbsp;</FONT></SPAN></DIV>
<DIV><SPAN class=857403607-16072002><FONT face=Arial color=#0000ff>For example, 
several ABNF errors were reported after bis-09</FONT></SPAN></DIV>
<DIV><SPAN class=857403607-16072002><FONT face=Arial color=#0000ff>and before 
RFC3261 which were fixed.&nbsp;&nbsp; I believe we will</FONT></SPAN></DIV>
<DIV><SPAN class=857403607-16072002><FONT face=Arial color=#0000ff>provide a 
diff to highlight the changes.&nbsp; This is harder than</FONT></SPAN></DIV>
<DIV><SPAN class=857403607-16072002><FONT face=Arial color=#0000ff>you might 
think because the RFC editor completely redid the</FONT></SPAN></DIV>
<DIV><SPAN class=857403607-16072002><FONT face=Arial color=#0000ff>formatting, 
so creating a diff is not really easy.&nbsp; </FONT></SPAN></DIV>
<DIV><SPAN class=857403607-16072002><FONT face=Arial 
color=#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=857403607-16072002><FONT face=Arial color=#0000ff>There are no 
actual </FONT></SPAN><SPAN class=857403607-16072002><FONT face=Arial 
color=#0000ff>protocol changes</FONT></SPAN><SPAN class=857403607-16072002><FONT 
face=Arial color=#0000ff>.</FONT></SPAN></DIV>
<DIV><SPAN class=857403607-16072002><FONT face=Arial 
color=#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=857403607-16072002><FONT face=Arial 
color=#0000ff>Brian</FONT></SPAN></DIV>
<BLOCKQUOTE 
style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid">
  <DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> ABDUL SATTAR SHAIK 
  [mailto:abdulss@migv.mot.com]<BR><B>Sent:</B> Tuesday, July 16, 2002 2:06 
  AM<BR><B>To:</B> 'sip@ietf.org'<BR><B>Subject:</B> [Sip] (no 
  subject)<BR><BR></FONT></DIV>
  <P><FONT face=Arial size=2>Hi</FONT> <BR><FONT face=Arial size=2>what are the 
  differences between rfc2543-bis-09 and rfc 3261?</FONT> </P>
  <P><FONT face=Arial size=2>I&nbsp; didn't find a single difference in these 
  two.</FONT> </P>
  <P><FONT face=Arial size=2>I think the bis-09 is&nbsp; published as 3261 since 
  bis-09 expires by august.</FONT> </P>
  <P><FONT face=Arial size=2>AmI right ?</FONT> </P>
  <P><FONT face=Arial size=2>Thanks</FONT> <BR><FONT face=Arial size=2>abdul 
  sattar</FONT> </P><BR></BLOCKQUOTE></BODY></HTML>

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From daemon@optimus.ietf.org  Tue Jul 16 05:45:09 2002
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From: Matthew.Gardiner@aculab.com
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Subject: [Sip] SIP call transfer as in draft-ietf-sipping-service-examples-02.tx
 t
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Hi All,

A colleague and I are currently trying to design call transfer into a system
and are therefore consulting draft-ietf-sipping-service-examples-02.txt to
assist us. We are finding the example of "Attended transfer" a little
confusing; we can understand messages F1 to F14, but we do not understand
how message F15 is able to confer to user A any useful information.

1) At this stage user A has a call held (id=12345600@here.com) with user B,
but the REFER (F15) message uses a new callid (12345678900@here.com).
2) Furthermore the REFER message itself does not seem to have any fields
which relate to the held call at user A.

If anyone understands how SIP call transfer is achieved, with regard to the
"Attended transfer" example in draft-ietf-sipping-service-examples-02.txt,
could they possibly send me an explanation?

Thanks for your time,

Matthew Gardiner
Software Engineer
Aculab 
Tel: +44 (0) 1908 273 911
Fax: +44 (0) 1908 273 801
Email: matthew.gardiner@aculab.com>
Website: http://www.aculab.com




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Subject: [Sip] SIP Processing power
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Dear all

How do you assess the processing power of SIP server ?
Is it based on the categorisation of the actions carried out according to
their complexities( simple,medium, complex) and then assigning a
corresponding processing power.
If so how you categorise an action as simple ,medium or complex.

Comments and answers are very much appreciated.

Regards

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From daemon@optimus.ietf.org  Tue Jul 16 05:52:29 2002
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From: "Shane McCormack" <smccormack@tssg.org>
To: "Dean Willis" <dean.willis@softarmor.com>,
        "'Avshalom Houri'" <AVSHALOM@il.ibm.com>, <sip@ietf.org>
Subject: RE: [Sip] The cost of UDP
Date: Tue, 16 Jul 2002 10:24:19 +0100
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Hi,
Would SCTP solve any resource issues here?

Shane

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Dean
Willis
Sent: 16 July 2002 08:01
To: 'Avshalom Houri'; sip@ietf.org
Subject: RE: [Sip] The cost of UDP


Avshalom said:
--------------
Wanted to remark on the remark that UDP is less resource consuming then
TCP and you can 
get more users on a machine with UDP. 

Well, it is very much dependent on your implementation. I have seen an
implementation that can 
hold 20-30K users on a simple NT machine quite easily. Saying that UDP
enable you to support 
more users is a bit misleading what happens is that you consume less
system resources but you 
end up in consuming much more application resources since now you need
to do what TCP does 
for you in the application level. Using TCP and TLS also goes with the
lines of using existing 
protocols to solve problems and not having to solve these problems again
and again in the 
applications. 
---------------

The "TCP limit" is based on the fact that most sockets-based
implementations of TCP use up one filesystem file handle per connected
socket. Since most kernel limitations use an unsigned 16-bit index for
the filehandle table, this puts a fairly hard upper bound of 64k
connected sockets. Given that one is likely to need a few to support the
OS and other applications, this makes a firm upper bound of around
64,000. Of course, many operating systems don't actually support this
many, with many TCP implementations actually breakikng at a much smaller
number, perhaps 16 or 256, but we're not likely to see commercial-grade
proxy systems built on these systems.


Of course, with connection-reuse in a proxy-proxy scenario, one socket
might actually be supporting thousands of users. The real issue is on
"boundary" proxies, where there is fan-in (or fan-out) to many UAs or
small (residential?) proxies.

What's the impact?

Let's say we average one 7-message dialog sequence per hour per device.
That's 7/messages per hour, or about 2 messages per second per 1,000 UA.
With an average proxy capable of handling 1,000 messages per second, we
should be able to handle 500,000 UA per proxy. That's clearly a larger
number than 64k, so there IS a potential for fan in/out limitations
being induced by TCP on larger servers.  Of course, if one factors in
presence and a little more traffic, it's not at all unreasoablle to
expect more than 70 messages per hour per UA, which with this size proxy
gets us CPU bound at around 50,000 clients, so the TCP constraint isn't
binding.

In other words, my analysis seems to indicate that for any "reasonable"
model of proxy fanout that the TCP connected socket limitation is only
to likely to impact really BIG proxy servers using a monolithic
"mainframe" style architecture not common in today's distributed
environments. In other words, it's just not much of a problem.

Another aspect to consider is the "half open connection limit". There
exists a constraint in many TCP implementations as to how many sockets
may be in a connecting (not connected) state at any given time. Just a
few years back, this number was something like 16 in a certain "robust
commercial UNIX". I think it's commonly much larger now. This was highly
problematic with HTTP 1.0 and it's very short-lived TCP connections, but
our model of persistent TCP/TLS connections and connection reuse greatly
reduce this liability.

Consider that 50,000 client server. If every single TCP connection is
broken and restored over the course of an hour, that gives us something
like 5 half/open connections at any given time. If however every
connection is restarted at exactly the same time, we DO have a bit more
of a problem, so there may well be an issue in this regard.

--
Dean


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From daemon@optimus.ietf.org  Tue Jul 16 08:47:19 2002
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CC: "'Avshalom Houri'" <AVSHALOM@il.ibm.com>, sip@ietf.org
Subject: Re: [Sip] The cost of UDP
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The "reboot building" restart issue is something we probably need to do 
something about in any event. Using TCP to the outbound proxy compounds 
the problem that already exists due to the REGISTER flood. But maybe all 
the devices will be fighting for DHCP IP addresses and thus spread out 
the onrush.

Dean Willis wrote:
> Avshalom said:
> --------------
> Wanted to remark on the remark that UDP is less resource consuming then
> TCP and you can 
> get more users on a machine with UDP. 
> 
> Well, it is very much dependent on your implementation. I have seen an
> implementation that can 
> hold 20-30K users on a simple NT machine quite easily. Saying that UDP
> enable you to support 
> more users is a bit misleading what happens is that you consume less
> system resources but you 
> end up in consuming much more application resources since now you need
> to do what TCP does 
> for you in the application level. Using TCP and TLS also goes with the
> lines of using existing 
> protocols to solve problems and not having to solve these problems again
> and again in the 
> applications. 
> ---------------
> 
> The "TCP limit" is based on the fact that most sockets-based
> implementations of TCP use up one filesystem file handle per connected
> socket. Since most kernel limitations use an unsigned 16-bit index for
> the filehandle table, this puts a fairly hard upper bound of 64k
> connected sockets. Given that one is likely to need a few to support the
> OS and other applications, this makes a firm upper bound of around
> 64,000. Of course, many operating systems don't actually support this
> many, with many TCP implementations actually breakikng at a much smaller
> number, perhaps 16 or 256, but we're not likely to see commercial-grade
> proxy systems built on these systems.
> 
> 
> Of course, with connection-reuse in a proxy-proxy scenario, one socket
> might actually be supporting thousands of users. The real issue is on
> "boundary" proxies, where there is fan-in (or fan-out) to many UAs or
> small (residential?) proxies.
> 
> What's the impact?
> 
> Let's say we average one 7-message dialog sequence per hour per device.
> That's 7/messages per hour, or about 2 messages per second per 1,000 UA.
> With an average proxy capable of handling 1,000 messages per second, we
> should be able to handle 500,000 UA per proxy. That's clearly a larger
> number than 64k, so there IS a potential for fan in/out limitations
> being induced by TCP on larger servers.  Of course, if one factors in
> presence and a little more traffic, it's not at all unreasoablle to
> expect more than 70 messages per hour per UA, which with this size proxy
> gets us CPU bound at around 50,000 clients, so the TCP constraint isn't
> binding.
> 
> In other words, my analysis seems to indicate that for any "reasonable"
> model of proxy fanout that the TCP connected socket limitation is only
> to likely to impact really BIG proxy servers using a monolithic
> "mainframe" style architecture not common in today's distributed
> environments. In other words, it's just not much of a problem.
> 
> Another aspect to consider is the "half open connection limit". There
> exists a constraint in many TCP implementations as to how many sockets
> may be in a connecting (not connected) state at any given time. Just a
> few years back, this number was something like 16 in a certain "robust
> commercial UNIX". I think it's commonly much larger now. This was highly
> problematic with HTTP 1.0 and it's very short-lived TCP connections, but
> our model of persistent TCP/TLS connections and connection reuse greatly
> reduce this liability.
> 
> Consider that 50,000 client server. If every single TCP connection is
> broken and restored over the course of an hour, that gives us something
> like 5 half/open connections at any given time. If however every
> connection is restarted at exactly the same time, we DO have a bit more
> of a problem, so there may well be an issue in this regard.
> 
> --
> Dean
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip



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From daemon@optimus.ietf.org  Tue Jul 16 09:08:45 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Shane McCormack'" <smccormack@tssg.org>,
        "'Avshalom Houri'" <AVSHALOM@il.ibm.com>, <sip@ietf.org>
Subject: RE: [Sip] The cost of UDP
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Shane asked:
> Hi,
> Would SCTP solve any resource issues here?
> 
> Shane

I don't know that we have enough deployment experience to really say.
Many people think so. A lot depends on the way that the SCTP library is
actually implemented, and I personally haven't implemented one yet . . .
Maybe somebody with implementation experience can say. Approaches that
follow the TCP Berkeley-sockets model of file-handle allocation would
presumably have the same problem. Other "raw socket" approaches might
not. Of course, one COULD presumably do a TCP implementation that isn't
file-handle bound, but I wouldn't recommend the one I did as a class
project n-many years ago as a starting point.

One place that SCTP should REALLY help is on optimizing high-volume
server-server links. TCP in this role has this annoying "head of line
blocking" quirk around retransmission of lost packets that SCTP fixes
nicely.


--
Dean


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From daemon@optimus.ietf.org  Tue Jul 16 21:05:28 2002
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Yes--this was pretty much the mood of the room in the SIMPLE wg meeting. 
It was also mentioned that this is a SIP draft, and any decisions must 
be made on the SIP list. (I am copying this message to SIP).

So, my current plan is to leave it alone. If others disagree, please 
express that on the SIP list.


Thanks!

Ben.


Cullen Jennings wrote:
> To quote the honorable Mr. Sparks "leave it alone"
> 
> I don't think that it is fundamentally broken (other than of course the lame
> name). Priority is definitely not the right thing to use. I would vote for
> just going with it as it is.
> 
> Cullen
> 
> 
> 
>>-----Original Message-----
>>From: simple-admin@mailman.dynamicsoft.com
>>[mailto:simple-admin@mailman.dynamicsoft.com]On Behalf Of Vijay K.
>>Gurbani
>>Sent: Tuesday, July 16, 2002 5:39 AM
>>To: bcampbell@dynamicsoft.com
>>Cc: simple@mailman.dynamicsoft.com
>>Subject: [Simple] -05 MESSAGE and Expires header
>>
>>
>>Ben:
>>
>>Any reasons why -05 MESSAGE I-D overloads "Expires: 0" to mean
>>immediate attention.  Maybe we can use "Priority: Urgent" instead; since
>>the Priority header is defined in 3261 and "may be factored into
>>decisions about call routing and acceptance".
>>
>>I think that overloading the Expires header here is confusing; 3261's
>>use of "Expires: 0" in conjunction with REGISTER means something
>>entirely different -- that this registration is invalid.  From a
>>semantic point of view, "Expires: 0" with REGISTER to mean
>>deregistration appears to make sense; but "Expires: 0" with MESSAGE to
>>mean immediate delivery appears incongrous.
>>
>>If it is not too late, maybe you can consider using the Priority header.
>>I won't be able to make it to simple WG meeting tomorrow, so I just
>>wanted to bring this to you attention.
>>
>>Thanks,
>>
>>- vijay
>>
>>_______________________________________________
>>simple mailing list
>>simple@mailman.dynamicsoft.com
>>http://mailman.dynamicsoft.com/mailman/listinfo/simple
>>
> 
> 




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From: "Cullen Jennings" <fluffy@cisco.com>
To: <sip@ietf.org>
Cc: "'Avshalom Houri'" <AVSHALOM@il.ibm.com>,
        "Dean Willis" <dean.willis@softarmor.com>
Subject: RE: [Sip] The cost of UDP
Date: Tue, 16 Jul 2002 17:55:47 -0700
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Few thoughts on this ....

64,000 subscriber per host is not very many - people that are going to
deploy millions of users don't want a boat load of servers to manage. 64,000
could correspond to around 50 cps. I can imagine proxies that easily do 500
cps on a cheap machine today and I can imagine proxies that do 30,000 cps on
a $1000 machine 10 years from now. Perhaps no one will care about that level
of performance but the point I am making is simple - 64,000 users per host
is very low.

If a host supported several IP address on a single interface, it might be
able to do more than 64k TCP connections. Should work in theory but probably
does not work in practice.

The following scenario is quite realistic - We end up with a city with
several million IP phones. This city regularly looses power resulting in all
UA simultaneously rebooting and trying to register.

Cullen


> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Dean
> Willis
> Sent: Tuesday, July 16, 2002 12:01 AM
> To: 'Avshalom Houri'; sip@ietf.org
> Subject: RE: [Sip] The cost of UDP
>
>
> Avshalom said:
> --------------
> Wanted to remark on the remark that UDP is less resource consuming then
> TCP and you can
> get more users on a machine with UDP.
>
> Well, it is very much dependent on your implementation. I have seen an
> implementation that can
> hold 20-30K users on a simple NT machine quite easily. Saying that UDP
> enable you to support
> more users is a bit misleading what happens is that you consume less
> system resources but you
> end up in consuming much more application resources since now you need
> to do what TCP does
> for you in the application level. Using TCP and TLS also goes with the
> lines of using existing
> protocols to solve problems and not having to solve these problems again
> and again in the
> applications.
> ---------------
>
> The "TCP limit" is based on the fact that most sockets-based
> implementations of TCP use up one filesystem file handle per connected
> socket. Since most kernel limitations use an unsigned 16-bit index for
> the filehandle table, this puts a fairly hard upper bound of 64k
> connected sockets. Given that one is likely to need a few to support the
> OS and other applications, this makes a firm upper bound of around
> 64,000. Of course, many operating systems don't actually support this
> many, with many TCP implementations actually breakikng at a much smaller
> number, perhaps 16 or 256, but we're not likely to see commercial-grade
> proxy systems built on these systems.
>
>
> Of course, with connection-reuse in a proxy-proxy scenario, one socket
> might actually be supporting thousands of users. The real issue is on
> "boundary" proxies, where there is fan-in (or fan-out) to many UAs or
> small (residential?) proxies.
>
> What's the impact?
>
> Let's say we average one 7-message dialog sequence per hour per device.
> That's 7/messages per hour, or about 2 messages per second per 1,000 UA.
> With an average proxy capable of handling 1,000 messages per second, we
> should be able to handle 500,000 UA per proxy. That's clearly a larger
> number than 64k, so there IS a potential for fan in/out limitations
> being induced by TCP on larger servers.  Of course, if one factors in
> presence and a little more traffic, it's not at all unreasoablle to
> expect more than 70 messages per hour per UA, which with this size proxy
> gets us CPU bound at around 50,000 clients, so the TCP constraint isn't
> binding.
>
> In other words, my analysis seems to indicate that for any "reasonable"
> model of proxy fanout that the TCP connected socket limitation is only
> to likely to impact really BIG proxy servers using a monolithic
> "mainframe" style architecture not common in today's distributed
> environments. In other words, it's just not much of a problem.
>
> Another aspect to consider is the "half open connection limit". There
> exists a constraint in many TCP implementations as to how many sockets
> may be in a connecting (not connected) state at any given time. Just a
> few years back, this number was something like 16 in a certain "robust
> commercial UNIX". I think it's commonly much larger now. This was highly
> problematic with HTTP 1.0 and it's very short-lived TCP connections, but
> our model of persistent TCP/TLS connections and connection reuse greatly
> reduce this liability.
>
> Consider that 50,000 client server. If every single TCP connection is
> broken and restored over the course of an hour, that gives us something
> like 5 half/open connections at any given time. If however every
> connection is restarted at exactly the same time, we DO have a bit more
> of a problem, so there may well be an issue in this regard.
>
> --
> Dean
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>


_______________________________________________
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From daemon@optimus.ietf.org  Tue Jul 16 22:01:12 2002
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From: "Shane McCormack" <smccormack@tssg.org>
To: "Dean Willis" <dean.willis@softarmor.com>,
        "'Shane McCormack'" <smccormack@tssg.org>,
        "'Avshalom Houri'" <AVSHALOM@il.ibm.com>, <sip@ietf.org>
Subject: RE: [Sip] The cost of UDP
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I agree, I know Coritel university in Italy has done some SIP SCTP work
using the lksctp open source linux kernel implementation of SCTP
http://sourceforge.net/projects/lksctp
and kphone/dissipate sip stack.
http://www.wirlab.net/kphone/

Maybe some more research should be done? or is being done?

Regards
Shane

-----Original Message-----
From: Dean Willis [mailto:dean.willis@softarmor.com]
Sent: 16 July 2002 13:43
To: 'Shane McCormack'; 'Avshalom Houri'; sip@ietf.org
Subject: RE: [Sip] The cost of UDP


*This message was transferred with a trial version of CommuniGate(tm) Pro*
Shane asked:
> Hi,
> Would SCTP solve any resource issues here?
>
> Shane

I don't know that we have enough deployment experience to really say.
Many people think so. A lot depends on the way that the SCTP library is
actually implemented, and I personally haven't implemented one yet . . .
Maybe somebody with implementation experience can say. Approaches that
follow the TCP Berkeley-sockets model of file-handle allocation would
presumably have the same problem. Other "raw socket" approaches might
not. Of course, one COULD presumably do a TCP implementation that isn't
file-handle bound, but I wouldn't recommend the one I did as a class
project n-many years ago as a starting point.

One place that SCTP should REALLY help is on optimizing high-volume
server-server links. TCP in this role has this annoying "head of line
blocking" quirk around retransmission of lost packets that SCTP fixes
nicely.


--
Dean



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Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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From daemon@optimus.ietf.org  Tue Jul 16 22:12:30 2002
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From: "Sp.Raja" <spraja@isofttechindia.com>
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Subject: [Sip] CANCEL FAQs
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Please help me out by answering these questions.

1. Can CANCEL be used to cancel a bye ?

2. Can CANCEL be send by the called entity (UAS) ? If yes then under what
circumstances

- Sp.Raja

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class=3D360232310-16072002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D360232310-16072002>1. Can =
CANCEL be=20
used to cancel a bye ? </SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D360232310-16072002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D360232310-16072002>2. Can =
CANCEL be=20
send by the called entity (UAS) ? If yes then under what=20
circumstances</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D360232310-16072002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D360232310-16072002>-=20
Sp.Raja</SPAN></FONT></DIV></BODY></HTML>

------=_NextPart_000_0000_01C22CE1.39DDDBE0--



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From daemon@optimus.ietf.org  Tue Jul 16 23:56:18 2002
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From: "Medhavi Bhatia" <mbhatia@nextone.com>
To: <sip@ietf.org>
References: <NGBBILMLKLEIKNHLGJILOEHHCCAA.spraja@isofttechindia.com>
Subject: Re: [Sip] CANCEL FAQs
Date: Tue, 16 Jul 2002 23:50:46 -0400
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This stuff generally belongs to the implementor's list: sip-implementors
<sip-implementors@cs.columbia.edu>

Generally CANCEL should not be used for transactions which are bound to
complete
quickly, and never for transactions for which you havent received a
provisional (1xx)
response. So the answer for (1) would be: probably not. Also the UAS cannot
cancel
a transaction started by the UAC (I guess that's what you meant), so it
cannot generate
the CANCEL ever.

----- Original Message -----
From: "Sp.Raja" <spraja@isofttechindia.com>
To: <sip@ietf.org>
Sent: Tuesday, July 16, 2002 6:25 AM
Subject: [Sip] CANCEL FAQs


> Please help me out by answering these questions.
>
> 1. Can CANCEL be used to cancel a bye ?
>
> 2. Can CANCEL be send by the called entity (UAS) ? If yes then under what
> circumstances
>
> - Sp.Raja
>


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From: "Medhavi Bhatia" <mbhatia@nextone.com>
To: =?utf-8?B?66WY642V7Je0?= <yoodl87@tta.or.kr>, <sip@ietf.org>
References: <022601c22cac$b9452a30$041b3dd3@LocalHost>
Subject: Re: [Sip] Please, Give me the answer to "bis"
Date: Tue, 16 Jul 2002 23:34:48 -0400
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A second version ;)

----- Original Message -----
From: "ëĽëě´" <yoodl87@tta.or.kr>
To: <sip@ietf.org>
Sent: Tuesday, July 16, 2002 5:39 AM
Subject: [Sip] Please, Give me the answer to "bis"


> Dear,
>
> I'm now in charge of testing of VoIP systems in TTA, which is 3rd party
testing authority about IT equipments in Korea.
> What I want to know is just about the meaning of "bis" in "RFC
2543bis-09".
>
> Sir, do you happen to know about "bis"? Please give me a hand?
>
> I'd like to get the answer from you  : Abbreviation of "bis" or the real
meaning of "bis"
>
>
> Yours sincerely!
>
>
>
>
> Duk-yell Yoo
> Tel : +82-31-724-0164
> e-Mail : yoodl87@tta.or.kr
>
> Engineering Staff
> Open Testbed team
> Network Test Center
> IT Testing Laboratory
> TTA(Telecommunications Technology Association), Republic of Korea
>


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From daemon@optimus.ietf.org  Wed Jul 17 03:10:21 2002
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Date: Wed, 17 Jul 2002 02:44:56 -0400
From: Henning Schulzrinne <hgs@cs.columbia.edu>
Organization: Columbia University
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Subject: [Sip] Discussion meeting for hearing-impaired issues for SIP
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Room 413, 17:30 - 18:30

Meeting is open; agenda is to identify concrete items that need to be 
done, based on the requirements document (and preferably identify 
candidate volunteers).

Henning Schulzrinne


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From daemon@optimus.ietf.org  Wed Jul 17 05:55:56 2002
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Subject: Re: [Sip] Anounce WGLC for draft-ietf-sip-callerpref
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Hi,

I have a couple of small comments on the draft.


Q1: My first comment (which I also gave a long time ago, when the first
version of the draft was released):

I think a usefull feature would be when a proxy modifies/inserts/removes
caller preft, recieved in messages from a client (or another proxy). A UA
may not know which preferences it's "allowed" to register/use, so the proxy
(using whatever policy) could change those.

The draft talks about "caller preferences vs CPL (or similar)" in proxies,
but this feature could in many cases be a combination of those, since the
proxy needs some kind of policy what/how/if to change in the caller
preference headers of the received message.

No, the draft really does NOT forbid proxies to do this, but I think it
would show more usefulness of the draft if is was described in chapter 6.4
(unless chapter 6.3 is valid also for the "UAC part" of proxies).


Q2: My second comment is about a caller prefs feature I've thought could be
included:

Would it be possible for a client to indicate if it supports (or, "wishes
to receive") early media or not (note: this proposal is NOT to start the
early vs local ringback discussion again!)? A forking proxy could eg set
different values for different legs, to prevent early media mixing (the
proxy is not allowed to change the media mode indicator to inactive in the
SDP, so...) at the caller node.

Regards,

Christer Holmberg
Ericsson Finland




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From daemon@optimus.ietf.org  Wed Jul 17 05:57:40 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
To: "=?iso-8859-1?B?J7f5tPa/rSc=?=" <yoodl87@tta.or.kr>, <sip@ietf.org>
Subject: RE: [Sip] Please, Give me the answer to "bis"
Date: Wed, 17 Jul 2002 04:15:00 -0500
Message-ID: <005001c22d72$6e91dd50$7e4d5d85@TXDWILLIS2>
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Literally, German for the English word "to", but often meaning
"between". One will see store hours posted in terms like "9 bis 18" It
has been common in ITU to use the suffix "bis" on a specification to
indicate an interim version for as long as I can remember.
=20
--
Dean
=20

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of =
=B7=F9=B4=F6=BF=AD
Sent: Tuesday, July 16, 2002 4:40 AM
To: sip@ietf.org
Subject: [Sip] Please, Give me the answer to "bis"




Dear,=20
=20
I'm now in charge of testing of VoIP systems in TTA, which is 3rd party
testing authority about IT equipments in Korea.
What I want to know is just about the meaning of "bis" in "RFC
2543bis-09".
=20
Sir, do you happen to know about "bis"? Please give me a hand?
=20
I'd like to get the answer from you  : Abbreviation of "bis" or the real
meaning of "bis"
=20
=20
Yours sincerely!
=20
=20
=20
=20
Duk-yell Yoo
Tel : +82-31-724-0164
e-Mail : yoodl87@tta.or.kr
=20
Engineering Staff
Open Testbed team
Network Test Center
IT Testing Laboratory=20
TTA(Telecommunications Technology Association), Republic of Korea


------=_NextPart_000_0051_01C22D48.85BBD550
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	charset="iso-8859-1"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<TITLE>Message</TITLE>

<META content=3D"MSHTML 6.00.2716.2200" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><SPAN class=3D905330909-17072002><FONT face=3DArial color=3D#0000ff =

size=3D2>Literally, German for the English word "to", but often meaning =
"between".=20
One will see store hours posted in terms like "9 bis 18" It has been =
common in=20
ITU to use the suffix "bis" on a specification to indicate&nbsp;an =
interim=20
version for as long as I can remember.</FONT></SPAN></DIV>
<DIV><SPAN class=3D905330909-17072002><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D905330909-17072002><FONT face=3DArial color=3D#0000ff =

size=3D2>--</FONT></SPAN></DIV>
<DIV><SPAN class=3D905330909-17072002><FONT face=3DArial color=3D#0000ff =

size=3D2>Dean</FONT></SPAN></DIV>
<DIV><SPAN class=3D905330909-17072002></SPAN>&nbsp;</DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV></DIV>
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr =
align=3Dleft><FONT=20
  face=3DTahoma size=3D2>-----Original Message-----<BR><B>From:</B>=20
  sip-admin@ietf.org [mailto:sip-admin@ietf.org] <B>On Behalf Of=20
  </B>=B7=F9=B4=F6=BF&shy;<BR><B>Sent:</B> Tuesday, July 16, 2002 4:40 =
AM<BR><B>To:</B>=20
  sip@ietf.org<BR><B>Subject:</B> [Sip] Please, Give me the answer to=20
  "bis"<BR><BR></FONT></DIV>
  <DIV><FONT size=3D2>
  <DIV><FONT size=3D2>
  <DIV><FONT size=3D2>Dear, </FONT></DIV>
  <DIV>&nbsp;</DIV>
  <DIV><FONT size=3D2>I'm now in charge of testing of VoIP systems in =
TTA, which=20
  is 3rd party testing authority&nbsp;about IT equipments in =
Korea.</FONT></DIV>
  <DIV><FONT size=3D2>What I&nbsp;want to know is just about the meaning =
of "bis"=20
  in "RFC 2543bis-09".</FONT></DIV>
  <DIV>&nbsp;</DIV>
  <DIV><FONT size=3D2>Sir,&nbsp;do you happen to know</FONT>&nbsp;<FONT=20
  size=3D2>about "bis"? Please give me a hand?</FONT></DIV>
  <DIV>&nbsp;</DIV>
  <DIV><FONT size=3D2>I'd like to get the answer from you&nbsp; : =
Abbreviation of=20
  "bis" or the real meaning of "bis"</FONT></DIV>
  <DIV>&nbsp;</DIV>
  <DIV><FONT size=3D2></FONT>&nbsp;</DIV>
  <DIV><FONT size=3D2>Yours sincerely!</FONT></DIV>
  <DIV>&nbsp;</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>&nbsp;</DIV>
  <DIV><FONT size=3D2></FONT>&nbsp;</DIV>
  <DIV><FONT size=3D2>Duk-yell Yoo</FONT></DIV>
  <DIV><FONT size=3D2>Tel : +82-31-724-0164</FONT></DIV>
  <DIV><FONT size=3D2>e-Mail : <A=20
  href=3D"mailto:yoodl87@tta.or.kr">yoodl87@tta.or.kr</A></FONT></DIV>
  <DIV>&nbsp;</DIV>
  <DIV><FONT size=3D2>Engineering&nbsp;Staff</FONT></DIV>
  <DIV><FONT size=3D2>Open Testbed team</FONT></DIV>
  <DIV><FONT size=3D2>Network Test Center</FONT></DIV>
  <DIV><FONT size=3D2>IT Testing Laboratory </FONT></DIV>
  <DIV><FONT size=3D2>TTA(Telecommunications Technology Association), =
Republic of=20
  =
Korea</FONT></DIV></FONT></DIV></FONT></DIV></BLOCKQUOTE></BODY></HTML>

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From daemon@optimus.ietf.org  Wed Jul 17 06:01:08 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Cullen Jennings'" <fluffy@cisco.com>, <sip@ietf.org>
Cc: "'Avshalom Houri'" <AVSHALOM@il.ibm.com>
Subject: RE: [Sip] The cost of UDP
Date: Wed, 17 Jul 2002 04:15:00 -0500
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> If a host supported several IP address on a single interface, 
> it might be able to do more than 64k TCP connections. Should 
> work in theory but probably does not work in practice.

Nope. It is a kernel limitation on the number of file handles available,
as I understand it.

--
Dean


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To: dean.willis@softarmor.com, yoodl87@tta.or.kr, sip@ietf.org
Subject: RE: [Sip] Please, Give me the answer to "bis"
Date: Wed, 17 Jul 2002 11:22:56 +0100
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I thought...
=20
'Bis' came from some latin or greek or something meaning 2nd.  And Ter =
means
3rd...
=20
=20

-----Original Message-----
From: Dean Willis [mailto:dean.willis@softarmor.com]
Sent: 17 July 2002 10:15
To: '=B7=F9=B4=F6=BF=AD'; sip@ietf.org
Subject: RE: [Sip] Please, Give me the answer to "bis"


Literally, German for the English word "to", but often meaning =
"between".
One will see store hours posted in terms like "9 bis 18" It has been =
common
in ITU to use the suffix "bis" on a specification to indicate an =
interim
version for as long as I can remember.
=20
--
Dean
=20

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of =
=B7=F9=B4=F6=BF=AD
Sent: Tuesday, July 16, 2002 4:40 AM
To: sip@ietf.org
Subject: [Sip] Please, Give me the answer to "bis"




Dear,=20
=20
I'm now in charge of testing of VoIP systems in TTA, which is 3rd party
testing authority about IT equipments in Korea.
What I want to know is just about the meaning of "bis" in "RFC =
2543bis-09".
=20
Sir, do you happen to know about "bis"? Please give me a hand?
=20
I'd like to get the answer from you  : Abbreviation of "bis" or the =
real
meaning of "bis"
=20
=20
Yours sincerely!
=20
=20
=20
=20
Duk-yell Yoo
Tel : +82-31-724-0164
e-Mail : yoodl87@tta.or.kr <mailto:yoodl87@tta.or.kr>=20
=20
Engineering Staff
Open Testbed team
Network Test Center
IT Testing Laboratory=20
TTA(Telecommunications Technology Association), Republic of Korea


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3DISO-8859-1">
<TITLE>Message</TITLE>

<META content=3D"MSHTML 5.00.3103.1000" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN =
class=3D183002410-17072002>I=20
thought...</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D183002410-17072002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN =
class=3D183002410-17072002>'Bis'=20
came from some latin or greek or something meaning 2nd.&nbsp; And Ter =
means=20
3rd...</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D183002410-17072002></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D183002410-17072002></SPAN></FONT>&nbsp;</DIV>
<BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
  <DIV align=3Dleft class=3DOutlookMessageHeader dir=3Dltr><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> Dean Willis=20
  [mailto:dean.willis@softarmor.com]<BR><B>Sent:</B> 17 July 2002=20
  10:15<BR><B>To:</B> '=B7=F9=B4=F6=BF&shy;'; =
sip@ietf.org<BR><B>Subject:</B> RE: [Sip]=20
  Please, Give me the answer to "bis"<BR><BR></DIV></FONT>
  <DIV><SPAN class=3D905330909-17072002><FONT color=3D#0000ff =
face=3DArial=20
  size=3D2>Literally, German for the English word "to", but often =
meaning=20
  "between". One will see store hours posted in terms like "9 bis 18" =
It has=20
  been common in ITU to use the suffix "bis" on a specification to=20
  indicate&nbsp;an interim version for as long as I can=20
  remember.</FONT></SPAN></DIV>
  <DIV><SPAN class=3D905330909-17072002><FONT color=3D#0000ff =
face=3DArial=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D905330909-17072002><FONT color=3D#0000ff =
face=3DArial=20
  size=3D2>--</FONT></SPAN></DIV>
  <DIV><SPAN class=3D905330909-17072002><FONT color=3D#0000ff =
face=3DArial=20
  size=3D2>Dean</FONT></SPAN></DIV>
  <DIV><SPAN class=3D905330909-17072002></SPAN>&nbsp;</DIV>
  <BLOCKQUOTE dir=3Dltr=20
  style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; =
MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">
    <DIV></DIV>
    <DIV align=3Dleft class=3DOutlookMessageHeader dir=3Dltr =
lang=3Den-us><FONT=20
    face=3DTahoma size=3D2>-----Original Message-----<BR><B>From:</B>=20
    sip-admin@ietf.org [mailto:sip-admin@ietf.org] <B>On Behalf Of=20
    </B>=B7=F9=B4=F6=BF&shy;<BR><B>Sent:</B> Tuesday, July 16, 2002 =
4:40 AM<BR><B>To:</B>=20
    sip@ietf.org<BR><B>Subject:</B> [Sip] Please, Give me the answer to =

    "bis"<BR><BR></FONT></DIV>
    <DIV><FONT size=3D2>
    <DIV><FONT size=3D2>
    <DIV><FONT size=3D2>Dear, </FONT></DIV>
    <DIV>&nbsp;</DIV>
    <DIV><FONT size=3D2>I'm now in charge of testing of VoIP systems in =
TTA, which=20
    is 3rd party testing authority&nbsp;about IT equipments in=20
    Korea.</FONT></DIV>
    <DIV><FONT size=3D2>What I&nbsp;want to know is just about the =
meaning of=20
    "bis" in "RFC 2543bis-09".</FONT></DIV>
    <DIV>&nbsp;</DIV>
    <DIV><FONT size=3D2>Sir,&nbsp;do you happen to =
know</FONT>&nbsp;<FONT=20
    size=3D2>about "bis"? Please give me a hand?</FONT></DIV>
    <DIV>&nbsp;</DIV>
    <DIV><FONT size=3D2>I'd like to get the answer from you&nbsp; : =
Abbreviation=20
    of "bis" or the real meaning of "bis"</FONT></DIV>
    <DIV>&nbsp;</DIV>
    <DIV><FONT size=3D2></FONT>&nbsp;</DIV>
    <DIV><FONT size=3D2>Yours sincerely!</FONT></DIV>
    <DIV>&nbsp;</DIV>
    <DIV>&nbsp;</DIV>
    <DIV>&nbsp;</DIV>
    <DIV><FONT size=3D2></FONT>&nbsp;</DIV>
    <DIV><FONT size=3D2>Duk-yell Yoo</FONT></DIV>
    <DIV><FONT size=3D2>Tel : +82-31-724-0164</FONT></DIV>
    <DIV><FONT size=3D2>e-Mail : <A=20
    =
href=3D"mailto:yoodl87@tta.or.kr">yoodl87@tta.or.kr</A></FONT></DIV>
    <DIV>&nbsp;</DIV>
    <DIV><FONT size=3D2>Engineering&nbsp;Staff</FONT></DIV>
    <DIV><FONT size=3D2>Open Testbed team</FONT></DIV>
    <DIV><FONT size=3D2>Network Test Center</FONT></DIV>
    <DIV><FONT size=3D2>IT Testing Laboratory </FONT></DIV>
    <DIV><FONT size=3D2>TTA(Telecommunications Technology Association), =
Republic=20
    of=20
Korea</FONT></DIV></FONT></DIV></FONT></DIV></BLOCKQUOTE></BLOCKQUOTE></=
BODY></HTML>

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From daemon@optimus.ietf.org  Wed Jul 17 07:22:36 2002
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Date: Wed, 17 Jul 2002 06:43:22 -0400
From: Henning Schulzrinne <hgs@cs.columbia.edu>
Organization: Columbia University
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To: duncan.mills@vf.vodafone.co.uk
CC: dean.willis@softarmor.com, yoodl87@tta.or.kr, sip@ietf.org
Subject: Re: [Sip] Please, Give me the answer to "bis"
References: <DDC3D921FB67D211AAD100A0C9E58F530B2CA37A@hampstead.vfl.vodafone>
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Merriam Webster says:

Main Entry: bis
Pronunciation: 'bis
Function: adverb
Etymology: Latin, from Old Latin dvis; akin to Old High German zwiro 
twice, Latin duo two -- more at TWO

Main Entry: ter-
Function: combining form
Etymology: Latin, from ter; akin to Greek & Sanskrit tris three times, 
Latin tres three -- more at THREE
: three times : threefold : three <tercentenary>

duncan.mills@vf.vodafone.co.uk wrote:
> I thought...
>  
> 'Bis' came from some latin or greek or something meaning 2nd.  And Ter means
> 3rd...
>  
>  
> 
> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: 17 July 2002 10:15
> To: 'ˇů´öż­'; sip@ietf.org
> Subject: RE: [Sip] Please, Give me the answer to "bis"
> 
> 
> Literally, German for the English word "to", but often meaning "between".
> One will see store hours posted in terms like "9 bis 18" It has been common
> in ITU to use the suffix "bis" on a specification to indicate an interim
> version for as long as I can remember.
>  
> --
> Dean
>  
> 
> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of ˇů´öż­
> Sent: Tuesday, July 16, 2002 4:40 AM
> To: sip@ietf.org
> Subject: [Sip] Please, Give me the answer to "bis"
> 
> 
> 
> 
> Dear, 
>  
> I'm now in charge of testing of VoIP systems in TTA, which is 3rd party
> testing authority about IT equipments in Korea.
> What I want to know is just about the meaning of "bis" in "RFC 2543bis-09".
>  
> Sir, do you happen to know about "bis"? Please give me a hand?
>  
> I'd like to get the answer from you  : Abbreviation of "bis" or the real
> meaning of "bis"
>  
>  
> Yours sincerely!
>  
>  
>  
>  
> Duk-yell Yoo
> Tel : +82-31-724-0164
> e-Mail : yoodl87@tta.or.kr <mailto:yoodl87@tta.or.kr> 
>  
> Engineering Staff
> Open Testbed team
> Network Test Center
> IT Testing Laboratory 
> TTA(Telecommunications Technology Association), Republic of Korea
> 
> 



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From daemon@optimus.ietf.org  Wed Jul 17 09:46:08 2002
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From: "Suresh Arora" <suresh.arora@baypackets.com>
To: "Medhavi Bhatia" <mbhatia@nextone.com>, <sip@ietf.org>
Subject: RE: [Sip] CANCEL FAQs
Date: Wed, 17 Jul 2002 18:14:33 +0530
Message-ID: <NGBBJFFHPANCFJENNHFDOEDOCDAA.suresh.arora@baypackets.com>
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I think UAS can send CANCEL to UAC for a re-INVITE (if requested session
characteristics are not supported by Server).

-Suresh

-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Medhavi
Bhatia
Sent: Wednesday, July 17, 2002 9:21 AM
To: sip@ietf.org
Subject: Re: [Sip] CANCEL FAQs


This stuff generally belongs to the implementor's list: sip-implementors
<sip-implementors@cs.columbia.edu>

Generally CANCEL should not be used for transactions which are bound to
complete
quickly, and never for transactions for which you havent received a
provisional (1xx)
response. So the answer for (1) would be: probably not. Also the UAS cannot
cancel
a transaction started by the UAC (I guess that's what you meant), so it
cannot generate
the CANCEL ever.

----- Original Message -----
From: "Sp.Raja" <spraja@isofttechindia.com>
To: <sip@ietf.org>
Sent: Tuesday, July 16, 2002 6:25 AM
Subject: [Sip] CANCEL FAQs


> Please help me out by answering these questions.
>
> 1. Can CANCEL be used to cancel a bye ?
>
> 2. Can CANCEL be send by the called entity (UAS) ? If yes then under what
> circumstances
>
> - Sp.Raja
>


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From daemon@optimus.ietf.org  Wed Jul 17 10:26:19 2002
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Hello All

The UPDATE method was defined for the dialog independent control of the session. Would it not be necessary to define an equivalent provisional response, that would allow to control the session without impacting the dialog state?

As currently the 183 response is widely used for this proposes and it's actually not clean in terms of call and bearer control separation.
 
Greeting,
Denis Alexeitsev 

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To: "Dean Willis" <dean.willis@softarmor.com>
Cc: "'Cullen Jennings'" <fluffy@cisco.com>, <sip@ietf.org>,
        "'Avshalom Houri'" <AVSHALOM@il.ibm.com>
Subject: RE: [Sip] The cost of UDP
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Dean Willis writes:
 > > If a host supported several IP address on a single interface, 
 > > it might be able to do more than 64k TCP connections. Should 
 > > work in theory but probably does not work in practice.
 > 
 > Nope. It is a kernel limitation on the number of file handles available,
 > as I understand it.

This doesn't really strike me as a reason to avoid
something. It may be crufty to change, but there
is nothing sacrosanct about 16 bit file
handles. There are *lots* of services where you'd
like to pile as many occassional users as possible
onto the server. IM comes to mind here too.

That said, I think that this entire discussion is
somewhat missing the point: how does post dial
delay fare with all things considered (e2e, qos,
security) with TCP and UDP? A self-contained
signed UDP exchange (kerberos, SMIME...) would
take 2 messages (invite/response). The equivalent
with TCP and TLS would take 10 messages (3, 5, 2).
This is in the realm of IKE la-la land. It's
probably even gnarlier with QoS preconditions.

	      Mike

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From daemon@optimus.ietf.org  Wed Jul 17 18:15:41 2002
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From: "Medhavi Bhatia" <mbhatia@nextone.com>
To: "Michael Thomas" <mat@cisco.com>,
        "Dean Willis" <dean.willis@softarmor.com>
Cc: "'Cullen Jennings'" <fluffy@cisco.com>, <sip@ietf.org>,
        "'Avshalom Houri'" <AVSHALOM@il.ibm.com>
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Subject: Re: [Sip] The cost of UDP
Date: Wed, 17 Jul 2002 17:59:21 -0400
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I dont think it comes from file descriptors. There are
several packages out there, which keep on stretching this limit.
Moreover I assume one file descriptors will be 64 bits.

I think its the TCP port size in the protocol, 16 bits.

----- Original Message -----
From: "Michael Thomas" <mat@cisco.com>
To: "Dean Willis" <dean.willis@softarmor.com>
Cc: "'Cullen Jennings'" <fluffy@cisco.com>; <sip@ietf.org>; "'Avshalom
Houri'" <AVSHALOM@il.ibm.com>
Sent: Wednesday, July 17, 2002 11:54 AM
Subject: RE: [Sip] The cost of UDP


> Dean Willis writes:
>  > > If a host supported several IP address on a single interface,
>  > > it might be able to do more than 64k TCP connections. Should
>  > > work in theory but probably does not work in practice.
>  >
>  > Nope. It is a kernel limitation on the number of file handles
available,
>  > as I understand it.
>
> This doesn't really strike me as a reason to avoid
> something. It may be crufty to change, but there
> is nothing sacrosanct about 16 bit file
> handles. There are *lots* of services where you'd
> like to pile as many occassional users as possible
> onto the server. IM comes to mind here too.
>
> That said, I think that this entire discussion is
> somewhat missing the point: how does post dial
> delay fare with all things considered (e2e, qos,
> security) with TCP and UDP? A self-contained
> signed UDP exchange (kerberos, SMIME...) would
> take 2 messages (invite/response). The equivalent
> with TCP and TLS would take 10 messages (3, 5, 2).
> This is in the realm of IKE la-la land. It's
> probably even gnarlier with QoS preconditions.
>
>       Mike
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>


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From daemon@ns.ietf.org  Wed Jul 17 21:08:50 2002
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From: "Medhavi Bhatia" <mbhatia@nextone.com>
To: "Michael Thomas" <mat@cisco.com>,
        "Dean Willis" <dean.willis@softarmor.com>,
        "Jiri Kuthan" <jiri@iptel.org>
Cc: "'Cullen Jennings'" <fluffy@cisco.com>, <sip@ietf.org>,
        "'Avshalom Houri'" <AVSHALOM@il.ibm.com>
References: <MFEJKLHKCMNFOPKPHMBPAEIBCCAA.fluffy@cisco.com> <005501c22d72$704cfad0$7e4d5d85@TXDWILLIS2> <15669.37709.46298.601697@thomasm-u1.cisco.com> <5.1.0.14.0.20020718004818.023552a0@iptel.org>
Subject: Re: [Sip] The cost of UDP
Date: Wed, 17 Jul 2002 20:56:28 -0400
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Several systems store the 32 bit descriptor in
a 16 bit quantity, like the sfio package from AT&T.

My thinking is that all these limits still come from the idea
of ports used in udp/tcp.

----- Original Message -----
From: "Jiri Kuthan" <jiri@iptel.org>
To: "Medhavi Bhatia" <mbhatia@nextone.com>; "Michael Thomas"
<mat@cisco.com>; "Dean Willis" <dean.willis@softarmor.com>
Cc: "'Cullen Jennings'" <fluffy@cisco.com>; <sip@ietf.org>; "'Avshalom
Houri'" <AVSHALOM@il.ibm.com>
Sent: Wednesday, July 17, 2002 6:49 PM
Subject: Re: [Sip] The cost of UDP


> TCP connections in operating systems I know are identified by
> quad-tupel {src_ip, dst_ip, src_port, dst_port}. Many more bits
> than just 16. If an operating system has 64k limitation on
> number of open TCP, than it is an implementation choice.
>
> -Jiri
>
> At 11:59 PM 7/17/2002, Medhavi Bhatia wrote:
> >I dont think it comes from file descriptors. There are
> >several packages out there, which keep on stretching this limit.
> >Moreover I assume one file descriptors will be 64 bits.
> >
> >I think its the TCP port size in the protocol, 16 bits.
> >
> >----- Original Message -----
> >From: "Michael Thomas" <mat@cisco.com>
> >To: "Dean Willis" <dean.willis@softarmor.com>
> >Cc: "'Cullen Jennings'" <fluffy@cisco.com>; <sip@ietf.org>; "'Avshalom
> >Houri'" <AVSHALOM@il.ibm.com>
> >Sent: Wednesday, July 17, 2002 11:54 AM
> >Subject: RE: [Sip] The cost of UDP
> >
> >
> >> Dean Willis writes:
> >>  > > If a host supported several IP address on a single interface,
> >>  > > it might be able to do more than 64k TCP connections. Should
> >>  > > work in theory but probably does not work in practice.
> >>  >
> >>  > Nope. It is a kernel limitation on the number of file handles
> >available,
> >>  > as I understand it.
> >>
> >> This doesn't really strike me as a reason to avoid
> >> something. It may be crufty to change, but there
> >> is nothing sacrosanct about 16 bit file
> >> handles. There are *lots* of services where you'd
> >> like to pile as many occassional users as possible
> >> onto the server. IM comes to mind here too.
> >>
> >> That said, I think that this entire discussion is
> >> somewhat missing the point: how does post dial
> >> delay fare with all things considered (e2e, qos,
> >> security) with TCP and UDP? A self-contained
> >> signed UDP exchange (kerberos, SMIME...) would
> >> take 2 messages (invite/response). The equivalent
> >> with TCP and TLS would take 10 messages (3, 5, 2).
> >> This is in the realm of IKE la-la land. It's
> >> probably even gnarlier with QoS preconditions.
> >>
> >>       Mike
> >>
> >> _______________________________________________
> >> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >> This list is for NEW development of the core SIP Protocol
> >> Use sip-implementors@cs.columbia.edu for questions on current sip
> >> Use sipping@ietf.org for new developments on the application of sip
> >>
> >
> >
> >_______________________________________________
> >Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >This list is for NEW development of the core SIP Protocol
> >Use sip-implementors@cs.columbia.edu for questions on current sip
> >Use sipping@ietf.org for new developments on the application of sip
>
>


_______________________________________________
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From daemon@ns.ietf.org  Wed Jul 17 21:19:48 2002
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Date: Wed, 17 Jul 2002 20:49:52 -0400
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To: Medhavi Bhatia <mbhatia@nextone.com>
CC: Michael Thomas <mat@cisco.com>, Dean Willis
 <dean.willis@softarmor.com>,
        Jiri Kuthan <jiri@iptel.org>, "'Cullen Jennings'"
 <fluffy@cisco.com>,
        sip@ietf.org, "'Avshalom Houri'"
 <AVSHALOM@il.ibm.com>
Subject: Re: [Sip] The cost of UDP
References: <MFEJKLHKCMNFOPKPHMBPAEIBCCAA.fluffy@cisco.com> <005501c22d72$704cfad0$7e4d5d85@TXDWILLIS2> <15669.37709.46298.601697@thomasm-u1.cisco.com> <5.1.0.14.0.20020718004818.023552a0@iptel.org> <015401c22df5$f39ff8a0$0100000a@nextone.com>
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Just in case there is confusion: The port limitation does not limit the 
number of concurrent connections. A TCP connection is identified by four 
quantities: source/destination port, source/destination address. Thus, 
there's no limit imposed by 16-bit ports unless you count the 
uninteresting case where all such connections go to the same destination 
address.

Medhavi Bhatia wrote:
> Several systems store the 32 bit descriptor in
> a 16 bit quantity, like the sfio package from AT&T.
> 
> My thinking is that all these limits still come from the idea
> of ports used in udp/tcp.
> 


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From daemon@ns.ietf.org  Wed Jul 17 22:03:44 2002
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To: "Medhavi Bhatia" <mbhatia@nextone.com>, "Michael Thomas" <mat@cisco.com>,
        "Dean Willis" <dean.willis@softarmor.com>
From: Jiri Kuthan <jiri@iptel.org>
Subject: Re: [Sip] The cost of UDP
Cc: "'Cullen Jennings'" <fluffy@cisco.com>, <sip@ietf.org>,
        "'Avshalom Houri'" <AVSHALOM@il.ibm.com>
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TCP connections in operating systems I know are identified by
quad-tupel {src_ip, dst_ip, src_port, dst_port}. Many more bits 
than just 16. If an operating system has 64k limitation on
number of open TCP, than it is an implementation choice.

-Jiri

At 11:59 PM 7/17/2002, Medhavi Bhatia wrote:
>I dont think it comes from file descriptors. There are
>several packages out there, which keep on stretching this limit.
>Moreover I assume one file descriptors will be 64 bits.
>
>I think its the TCP port size in the protocol, 16 bits.
>
>----- Original Message -----
>From: "Michael Thomas" <mat@cisco.com>
>To: "Dean Willis" <dean.willis@softarmor.com>
>Cc: "'Cullen Jennings'" <fluffy@cisco.com>; <sip@ietf.org>; "'Avshalom
>Houri'" <AVSHALOM@il.ibm.com>
>Sent: Wednesday, July 17, 2002 11:54 AM
>Subject: RE: [Sip] The cost of UDP
>
>
>> Dean Willis writes:
>>  > > If a host supported several IP address on a single interface,
>>  > > it might be able to do more than 64k TCP connections. Should
>>  > > work in theory but probably does not work in practice.
>>  >
>>  > Nope. It is a kernel limitation on the number of file handles
>available,
>>  > as I understand it.
>>
>> This doesn't really strike me as a reason to avoid
>> something. It may be crufty to change, but there
>> is nothing sacrosanct about 16 bit file
>> handles. There are *lots* of services where you'd
>> like to pile as many occassional users as possible
>> onto the server. IM comes to mind here too.
>>
>> That said, I think that this entire discussion is
>> somewhat missing the point: how does post dial
>> delay fare with all things considered (e2e, qos,
>> security) with TCP and UDP? A self-contained
>> signed UDP exchange (kerberos, SMIME...) would
>> take 2 messages (invite/response). The equivalent
>> with TCP and TLS would take 10 messages (3, 5, 2).
>> This is in the realm of IKE la-la land. It's
>> probably even gnarlier with QoS preconditions.
>>
>>       Mike
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>
>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip 



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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Michael Thomas'" <mat@cisco.com>
Cc: "'Cullen Jennings'" <fluffy@cisco.com>, <sip@ietf.org>,
        "'Avshalom Houri'" <AVSHALOM@il.ibm.com>
Subject: RE: [Sip] The cost of UDP
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> That said, I think that this entire discussion is
> somewhat missing the point: how does post dial
> delay fare with all things considered (e2e, qos,
> security) with TCP and UDP? A self-contained
> signed UDP exchange (kerberos, SMIME...) would
> take 2 messages (invite/response). The equivalent
> with TCP and TLS would take 10 messages (3, 5, 2).
> This is in the realm of IKE la-la land. It's
> probably even gnarlier with QoS preconditions.

Well, there is a little issue with fragmentation, so while only two
MESSAGES might be required, it could well be many PACKETS. And given any
probability of loss, a retry of an entire message occurs, many many more
packets. And of course there is always the unstated issue of key
management -- somewhere in there you also have to take into account the
talking to the ticket server and so on.

But Mike has a real point here -- the reality is much messier than the
simplfied models that often get discussed, and we probably do need to
make a habit of looking at a lot of these issues in more detail.

In my experience, proper security of any kind is non-trivial and often
considered prohibitively expensive. I believe this accounts for the
generally poor level of security we find in the real world. In other
words, experience dictates that some things may be worth doing, but NOT
doing right.

Thanks,

--
Dean


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From daemon@ns.ietf.org  Wed Jul 17 22:07:47 2002
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From: Kedar Patil <kpatil@sta.samsung.com>
To: "'Sp.Raja'" <spraja@isofttechindia.com>
Cc: sip@ietf.org
Subject: RE: [Sip] CANCEL FAQs
Date: Wed, 17 Jul 2002 12:11:22 -0500
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Hi,
 
(1) RFC 3261, section 9.1 :
 
=================================================================
   A CANCEL request SHOULD NOT be sent to cancel a request other than
   INVITE.
==========================================================
 
and 
 
=================================================================
If no provisional response has been received, the CANCEL request MUST
NOT be sent; rather, the client MUST wait for the arrival of a
provisional response before sending the request. 
=================================================================
 
Practically, even if you send CANCEL to BYE, 3 possibilities exist -
 
 - if it reaches before BYE, it will be rejected with 481.
 - if it reaches after BYE, UAS will process BYE first, i.e. it will send
final resp. to BYE first,
   then process CANCEL, which will rejected with 481.
 - if CANCEL to BYE is received by a stateful proxy server, it is a race
condition. ( I guess it
   can be so even with CANCEL to INVITE via a stateful proxy, is it?).
 
(2) Any UA that initiates a request is a UAC.
Now if called UA initiates INVITE (actually re-INVITE), then it should be
able to CANCEL it too.
 
thanks,
= Kedar =
 

-----Original Message-----
From: Sp.Raja [mailto:spraja@isofttechindia.com]
Sent: Tuesday, July 16, 2002 5:26 AM
To: sip@ietf.org
Subject: [Sip] CANCEL FAQs


Please help me out by answering these questions.
 
1. Can CANCEL be used to cancel a bye ? 
 
2. Can CANCEL be send by the called entity (UAS) ? If yes then under what
circumstances
 
- Sp.Raja


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<BODY>
<DIV><SPAN class=632524416-17072002><FONT face=Arial 
size=2>Hi,</FONT></SPAN></DIV>
<DIV><SPAN class=632524416-17072002><FONT face=Arial 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=632524416-17072002><FONT face=Arial size=2>(1) RFC 3261, 
section 9.1 :</FONT></SPAN></DIV>
<DIV><SPAN class=632524416-17072002><FONT face=Arial 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=632524416-17072002><FONT face=Arial 
size=2>=================================================================</FONT></SPAN></DIV>
<DIV><SPAN class=632524416-17072002><FONT face=Arial>&nbsp;&nbsp; A CANCEL 
request SHOULD NOT be sent to cancel a request other than<BR>&nbsp;&nbsp; 
INVITE.</FONT></SPAN></DIV>
<DIV><SPAN class=632524416-17072002><FONT 
face=Arial>==========================================================</FONT></SPAN></DIV>
<DIV><SPAN class=632524416-17072002><FONT face=Arial 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=632524416-17072002><FONT face=Arial size=2>and 
</FONT></SPAN></DIV>
<DIV><SPAN class=632524416-17072002><FONT face=Arial 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=632524416-17072002><FONT face=Arial 
size=2>=================================================================</FONT></SPAN></DIV>
<DIV><SPAN class=632524416-17072002><FONT face=Arial>If no provisional response 
has been received, the CANCEL request MUST<BR>NOT be sent; rather, the client 
MUST wait for the arrival of a<BR>provisional response before sending the 
request. </FONT></SPAN></DIV>
<DIV><SPAN class=632524416-17072002><FONT face=Arial 
size=2>=================================================================</FONT></SPAN></DIV>
<DIV><SPAN class=632524416-17072002><FONT face=Arial 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=632524416-17072002><FONT face=Arial size=2>Practically, even if 
you send CANCEL to BYE, 3 possibilities exist&nbsp;-</FONT></SPAN></DIV>
<DIV><SPAN class=632524416-17072002><FONT face=Arial 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=632524416-17072002><FONT face=Arial size=2>&nbsp;- if it 
reaches before BYE, it will be rejected with 481.</FONT></SPAN></DIV>
<DIV><SPAN class=632524416-17072002><FONT face=Arial size=2>&nbsp;- if it 
reaches after BYE, UAS will process BYE first, i.e. it will send final resp. to 
BYE first,</FONT></SPAN></DIV>
<DIV><SPAN class=632524416-17072002><FONT face=Arial 
size=2>&nbsp;&nbsp;&nbsp;then process CANCEL, which will rejected with 
481.</FONT></SPAN></DIV>
<DIV><SPAN class=632524416-17072002><FONT face=Arial size=2>&nbsp;- if CANCEL to 
BYE is received by a&nbsp;stateful proxy server, it&nbsp;is a race condition. ( 
I guess it</FONT></SPAN></DIV>
<DIV><SPAN class=632524416-17072002><FONT face=Arial 
size=2>&nbsp;&nbsp;&nbsp;can be&nbsp;so even </FONT></SPAN><SPAN 
class=632524416-17072002><FONT face=Arial size=2>with CANCEL to INVITE&nbsp;via 
a stateful proxy, is it?).</FONT></SPAN></DIV>
<DIV><SPAN class=632524416-17072002><FONT face=Arial 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=632524416-17072002><FONT face=Arial size=2>(2) Any UA&nbsp;that 
initiates a request is a UAC.</FONT></SPAN></DIV>
<DIV><SPAN class=632524416-17072002><FONT face=Arial size=2>Now if called UA 
initiates INVITE (actually re-INVITE), </FONT></SPAN><SPAN 
class=632524416-17072002><FONT face=Arial size=2>then it should be able to 
CANCEL it too.</FONT></SPAN></DIV>
<DIV><SPAN class=632524416-17072002><FONT face=Arial 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=632524416-17072002><FONT face=Arial 
size=2>thanks,</FONT></SPAN></DIV>
<DIV><SPAN class=632524416-17072002><FONT face=Arial size=2>= Kedar 
=</FONT></SPAN></DIV>
<DIV><SPAN class=632524416-17072002><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<BLOCKQUOTE dir=ltr 
style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px">
  <DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> Sp.Raja 
  [mailto:spraja@isofttechindia.com]<BR><B>Sent:</B> Tuesday, July 16, 2002 5:26 
  AM<BR><B>To:</B> sip@ietf.org<BR><B>Subject:</B> [Sip] CANCEL 
  FAQs<BR><BR></FONT></DIV>
  <DIV><FONT face=Arial size=2><SPAN class=360232310-16072002>Please help me out 
  by answering these questions.</SPAN></FONT></DIV>
  <DIV><FONT face=Arial size=2><SPAN 
  class=360232310-16072002></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=Arial size=2><SPAN class=360232310-16072002>1. Can CANCEL be 
  used to cancel a bye ? </SPAN></FONT></DIV>
  <DIV><FONT face=Arial size=2><SPAN 
  class=360232310-16072002></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=Arial size=2><SPAN class=360232310-16072002>2. Can CANCEL be 
  send by the called entity (UAS) ? If yes then under what 
  circumstances</SPAN></FONT></DIV>
  <DIV><FONT face=Arial size=2><SPAN 
  class=360232310-16072002></SPAN></FONT>&nbsp;</DIV>
  <DIV><FONT face=Arial size=2><SPAN class=360232310-16072002>- 
  Sp.Raja</SPAN></FONT></DIV></BLOCKQUOTE></BODY></HTML>

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From daemon@ns.ietf.org  Thu Jul 18 01:11:42 2002
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From: "Sp.Raja" <spraja@isofttechindia.com>
To: <sip@ietf.org>
Date: Thu, 18 Jul 2002 09:57:08 +0530
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Subject: [Sip] BYE race and re-INVITE doubt
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Hi,

Please clear my doubts on the following

1. What is expected out of the SIP element when it receives BYE after
sending a BYE?
Ex. Let UA1 and UA2 be two user agents and P1, P2 be proxies

       UA1    P1      P2      UA2
       |\      |       |      /|
       | \     |       |     / |
       |  \    |       |    /  |
       |   \   |       |   /   |
       |BYE \  |       |  / BYE|
       |     \>|\     /|</     |
       |       | \   / |       |
       |       |  \ /  |       |
       |       |   \   |       |
       |       |  / \  |       |
       |       *</   \>*       |
       |       |       |       |
       |       |       |       |

What will P1 and P2 do now after receiving BYE? They have already received
BYE from the opposite side and yet to receive the response for it.


2. When can re-INVITE be send?
I assume it can be send only after receiving ACK or sending ACK accordingly.
What is expected out of the proxy in the following cases? Does it simply
drops the re-INVITE?

UAC sending re-INVITE after sending INVITE and yet to receive final response
UAC sending re-INVITE instead of ACK after sending INVITE, received final
response
UAS sending re-INVITE after receiving INVITE
UAS sending re-INVITE after sending final response to INVITE and before
receiving ACK




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From daemon@ns.ietf.org  Thu Jul 18 01:15:10 2002
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From: "Ranjit Avasarala" <ranjit.avasarala@bplmail.com>
To: "Suresh Arora" <suresh.arora@baypackets.com>,
        "Medhavi Bhatia" <mbhatia@nextone.com>, <sip@ietf.org>
References: <NGBBJFFHPANCFJENNHFDOEDOCDAA.suresh.arora@baypackets.com>
Subject: Re: [Sip] CANCEL FAQs
Date: Thu, 18 Jul 2002 10:04:46 +0530
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u cannot send CANCEL for re-invite as u won't be getting any ringing tone as
response

Ranjit
----- Original Message -----
From: Suresh Arora <suresh.arora@baypackets.com>
To: Medhavi Bhatia <mbhatia@nextone.com>; <sip@ietf.org>
Sent: Wednesday, July 17, 2002 6:14 PM
Subject: RE: [Sip] CANCEL FAQs


> I think UAS can send CANCEL to UAC for a re-INVITE (if requested session
> characteristics are not supported by Server).
>
> -Suresh
>
> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Medhavi
> Bhatia
> Sent: Wednesday, July 17, 2002 9:21 AM
> To: sip@ietf.org
> Subject: Re: [Sip] CANCEL FAQs
>
>
> This stuff generally belongs to the implementor's list: sip-implementors
> <sip-implementors@cs.columbia.edu>
>
> Generally CANCEL should not be used for transactions which are bound to
> complete
> quickly, and never for transactions for which you havent received a
> provisional (1xx)
> response. So the answer for (1) would be: probably not. Also the UAS
cannot
> cancel
> a transaction started by the UAC (I guess that's what you meant), so it
> cannot generate
> the CANCEL ever.
>
> ----- Original Message -----
> From: "Sp.Raja" <spraja@isofttechindia.com>
> To: <sip@ietf.org>
> Sent: Tuesday, July 16, 2002 6:25 AM
> Subject: [Sip] CANCEL FAQs
>
>
> > Please help me out by answering these questions.
> >
> > 1. Can CANCEL be used to cancel a bye ?
> >
> > 2. Can CANCEL be send by the called entity (UAS) ? If yes then under
what
> > circumstances
> >
> > - Sp.Raja
> >
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From daemon@ns.ietf.org  Thu Jul 18 05:52:28 2002
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Subject: Re: [Sip] CANCEL FAQs
From: Robert Sparks <rsparks@dynamicsoft.com>
To: Ranjit Avasarala <ranjit.avasarala@bplmail.com>
Cc: Suresh Arora <suresh.arora@baypackets.com>,
        Medhavi Bhatia
	 <mbhatia@nextone.com>, sip@ietf.org
In-Reply-To: <002d01c22e14$71b7f4e0$5f0d0a0a@ranjit>
References: <NGBBJFFHPANCFJENNHFDOEDOCDAA.suresh.arora@baypackets.com> 
	<002d01c22e14$71b7f4e0$5f0d0a0a@ranjit>
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I'm going to respond to this on the sip-implementors list.

That is the right forum to ask these kind of questions (and to
answer them). This list focuses on the SIP working groups
current design issues.

If you haven't subscribed to sip-implementors (and are interested),
the information you need to do so can be found at
http://www.cs.columbia.edu/~hgs/sip/list.html

RjS

On Thu, 2002-07-18 at 13:34, Ranjit Avasarala wrote:
> u cannot send CANCEL for re-invite as u won't be getting any ringing tone as
> response
> 
> Ranjit
> ----- Original Message -----
> From: Suresh Arora <suresh.arora@baypackets.com>
> To: Medhavi Bhatia <mbhatia@nextone.com>; <sip@ietf.org>
> Sent: Wednesday, July 17, 2002 6:14 PM
> Subject: RE: [Sip] CANCEL FAQs
> 
> 
> > I think UAS can send CANCEL to UAC for a re-INVITE (if requested session
> > characteristics are not supported by Server).
> >
> > -Suresh
> >
> > -----Original Message-----
> > From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Medhavi
> > Bhatia
> > Sent: Wednesday, July 17, 2002 9:21 AM
> > To: sip@ietf.org
> > Subject: Re: [Sip] CANCEL FAQs
> >
> >
> > This stuff generally belongs to the implementor's list: sip-implementors
> > <sip-implementors@cs.columbia.edu>
> >
> > Generally CANCEL should not be used for transactions which are bound to
> > complete
> > quickly, and never for transactions for which you havent received a
> > provisional (1xx)
> > response. So the answer for (1) would be: probably not. Also the UAS
> cannot
> > cancel
> > a transaction started by the UAC (I guess that's what you meant), so it
> > cannot generate
> > the CANCEL ever.
> >
> > ----- Original Message -----
> > From: "Sp.Raja" <spraja@isofttechindia.com>
> > To: <sip@ietf.org>
> > Sent: Tuesday, July 16, 2002 6:25 AM
> > Subject: [Sip] CANCEL FAQs
> >
> >
> > > Please help me out by answering these questions.
> > >
> > > 1. Can CANCEL be used to cancel a bye ?
> > >
> > > 2. Can CANCEL be send by the called entity (UAS) ? If yes then under
> what
> > > circumstances
> > >
> > > - Sp.Raja
> > >
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip



_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From daemon@ns.ietf.org  Thu Jul 18 08:31:53 2002
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To: AC Mahendran <mahendra@qualcomm.com>
CC: sip@ietf.org, jari.arkko@lmf.ericsson.se, vesa.torvinen@lmf.ericsson.se
Subject: Re: [Sip] draft-ietf-sip-sec-agree-04.txt
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Hello,

In the particular example you are referring to, the server did not need
to include any header in the response in order to establish the chosen
security. However, in the general case, the server will chose which
information it should include in the response based on the
security-client header.

Modifying the security-client header can of course be used for a DoS
attack, but keep in mind that what we are trying to avoid is a bid-down
attack.

Regards,

Gonzalo

AC Mahendran wrote:
> 
> I have question regarding the purpose of the "Security-Client" field. Based
> on the description, it looks like this field is not used by the server in
> any way. Can someone explain the purpose of this field?
> 
> thanks,
> AC
> 
> Ps: In the "Server Initiated" (Section 3.4.2) procedures, this field is not
> used. By the same reasoning, this field should not be applicable to the
> "Client Initiated" procedures as well.
> 
> At 12:33 PM 7/7/2002 -0400, Internet-Drafts@ietf.org wrote:
> >A New Internet-Draft is available from the on-line Internet-Drafts
> >directories.
> >This draft is a work item of the Session Initiation Protocol Working Group
> >of the IETF.
> >
> >         Title           : Security Mechanism Agreement for SIP Sessions
> >         Author(s)       : J. Arkko et al.
> >         Filename        : draft-ietf-sip-sec-agree-04.txt
> >         Pages           : 17
> >         Date            : 05-Jul-02
> >
> >SIP has a number of security mechanisms. Some of them have been built
> >in to the SIP protocol, such as HTTP authentication or secure
> >attachments. These mechanisms have even alternative algorithms and
> >parameters. SIP does not currently provide any mechanism for
> >selecting which security mechanisms to use between two entities. In
> >particular, even if some mechanisms such as OPTIONS were used to make
> >this selection, the selection would be vulnerable against the
> >Bidding-Down attack. This document defines three header fields for
> >negotiating the security mechanisms within SIP between a SIP entity
> >and its next SIP hop. A SIP entity applying this mechanism must
> >always require some minimum security (i.e. integrity protection) from
> >all communicating parties in order to secure the negotiation, but the
> >negotiation can agree on which specific minimum security is used.
> >
> >A URL for this Internet-Draft is:
> >http://www.ietf.org/internet-drafts/draft-ietf-sip-sec-agree-04.txt
> >
> >To remove yourself from the IETF Announcement list, send a message to
> >ietf-announce-request with the word unsubscribe in the body of the message.
> >
> >Internet-Drafts are also available by anonymous FTP. Login with the username
> >"anonymous" and a password of your e-mail address. After logging in,
> >type "cd internet-drafts" and then
> >         "get draft-ietf-sip-sec-agree-04.txt".
> >
> >A list of Internet-Drafts directories can be found in
> >http://www.ietf.org/shadow.html
> >or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> >
> >
> >Internet-Drafts can also be obtained by e-mail.
> >
> >Send a message to:
> >         mailserv@ietf.org.
> >In the body type:
> >         "FILE /internet-drafts/draft-ietf-sip-sec-agree-04.txt".
> >
> >NOTE:   The mail server at ietf.org can return the document in
> >         MIME-encoded form by using the "mpack" utility.  To use this
> >         feature, insert the command "ENCODING mime" before the "FILE"
> >         command.  To decode the response(s), you will need "munpack" or
> >         a MIME-compliant mail reader.  Different MIME-compliant mail readers
> >         exhibit different behavior, especially when dealing with
> >         "multipart" MIME messages (i.e. documents which have been split
> >         up into multiple messages), so check your local documentation on
> >         how to manipulate these messages.
> >
> >
> >Below is the data which will enable a MIME compliant mail reader
> >implementation to automatically retrieve the ASCII version of the
> >Internet-Draft.
> >Content-Type: text/plain
> >Content-ID:     <20020705142509.I-D@ietf.org>
> >
> >ENCODING mime
> >FILE /internet-drafts/draft-ietf-sip-sec-agree-04.txt
> >
> ><ftp://ftp.ietf.org/internet-drafts/draft-ietf-sip-sec-agree-04.txt>
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo


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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From daemon@optimus.ietf.org  Fri Jul 19 07:56:30 2002
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From: "Matthias Konrad" <mkonrad@hsr.ch>
To: "Sp.Raja" <spraja@isofttechindia.com>, <sip@ietf.org>
References: <NGBBILMLKLEIKNHLGJILOEIBCCAA.spraja@isofttechindia.com>
Subject: Re: [Sip] BYE race and re-INVITE doubt
Date: Fri, 19 Jul 2002 13:14:58 +0200
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Hi,
I think that a stateful proxy doesn't care about any of the cases
you described. A proxy, as far as I know, shouldn't hold any dialog states.
He just knows about INVITE and nonINVITE transaction states. If that is
true, he doesnt even know, that the two BYE Requests belongs to the same
dialog.


----- Original Message -----
From: "Sp.Raja" <spraja@isofttechindia.com>
To: <sip@ietf.org>
Sent: Thursday, July 18, 2002 6:27 AM
Subject: [Sip] BYE race and re-INVITE doubt


> Hi,
>
> Please clear my doubts on the following
>
> 1. What is expected out of the SIP element when it receives BYE after
> sending a BYE?
> Ex. Let UA1 and UA2 be two user agents and P1, P2 be proxies
>
>        UA1    P1      P2      UA2
>        |\      |       |      /|
>        | \     |       |     / |
>        |  \    |       |    /  |
>        |   \   |       |   /   |
>        |BYE \  |       |  / BYE|
>        |     \>|\     /|</     |
>        |       | \   / |       |
>        |       |  \ /  |       |
>        |       |   \   |       |
>        |       |  / \  |       |
>        |       *</   \>*       |
>        |       |       |       |
>        |       |       |       |
>
> What will P1 and P2 do now after receiving BYE? They have already received
> BYE from the opposite side and yet to receive the response for it.
>
>
> 2. When can re-INVITE be send?
> I assume it can be send only after receiving ACK or sending ACK
accordingly.
> What is expected out of the proxy in the following cases? Does it simply
> drops the re-INVITE?
>
> UAC sending re-INVITE after sending INVITE and yet to receive final
response
> UAC sending re-INVITE instead of ACK after sending INVITE, received final
> response
> UAS sending re-INVITE after receiving INVITE
> UAS sending re-INVITE after sending final response to INVITE and before
> receiving ACK
>
>
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>




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From daemon@optimus.ietf.org  Fri Jul 19 09:25:57 2002
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From: Salvador Rey Calatayud <salreyca@teleco.upv.es>
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Hi everyone,

as for the transaction state diagrams, and section 14.1 of RFC 3261, a UAS that
had received an INVITE(not blank) cannot send another one, and thus start a new
offer/answer cycle, until it sends a 200ok, or receives the ACK for a 300-699
response. why is the rule different for 200ok than for other responses?

section 14.1 RFC 3261
"2. If there is an ongoing INVITE server transaction, the TU MUST wait until the
transaction reaches the confirmed or terminated state before initiating the new
INVITE."

I think UPDATE and this are not alligned since the UPDATE specification says
that another offer cannot be sent until a previous 1xxrel containing a
descriptor is acknowledged with a PRACK (playing the role of ACK to 200ok in
basic SIP).

I include section 5.1 of UPDATE-02
"and for the callee:
o If the UPDATE is being sent before the completion of the
INVITE transaction, and the initial INVITE contained an offer,
the UPDATE cannot be sent unless the callee has generated an
answer in a reliable provisional response, !!has received a
PRACK for that reliable provisional response!!, has not received
any requests (PRACK or UPDATE) with offers that it has not
answered, and has not sent any UPDATE requests containing
offers that have not been answered."

comments?

regards,
Salva

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From daemon@ns.ietf.org  Fri Jul 19 18:42:26 2002
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From: "Vijay K. Gurbani" <vkg@lucent.com>
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CC: SIP LIST <sip@ietf.org>, Vijay Gurbani <vkg@lucent.com>
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Subject: [Sip] Re: [Simple] -05 MESSAGE and Expires header
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This has been an interesting discussion with more people weighing
in the 'leave it alone' category.  However, here is an additional
point to consider: Expires is one of the headers that can be
encrypted.  An S/MIME encrypted MESSAGE request containing 'Expires: 0'
arriving at a relay UAS will get the same processing as will
an unencrypted MESSAGE with 'Expires: 3600' -- probably not good.  The
Priority header, on the other hand, is not permitted to be encrypted
(Table 3, RFC 3261); thus relays can make use of it.

Jonathan Rosenberg gave a concise explanation on why Priority and
'Expires: 0' to mean immediate delivery are different:

 > I believe that priority is quite different. A message can be for
 > immediate delivery, and be either urgent ("my house is on fire") or
 > not ("leaving now for lunch"). A message can be OK for storage (some
 > non-zero expires) and be urgent ("I need that business presentation
 > by 10pm!") or non-urgent ("Did you see that movie"). The fact that two
 > things can be used orthogonally is a sign that they are, in fact, two
 > different things.

In this context, maybe Priority is not the right header, but it is
the closest we have in base SIP to impart immediate delivery.

The whole point of 'Expires: 0' in the MESSAGE I-D is for the relay UAS
and the destination UAS to act on this request immediately.  Since the
Expire header can be encrypted, the directive of immediate delivery is
lost at the relay UAS.  Priority preserves it, and in fact, to avoid any
further ambiguity,  the MESSAGE I-D can further restrict the presence of
the Priority header in only those IM messages that denote urgency of
some sort, and thus need special processing by the relays.

Regards,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and Services
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216



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Section 17.2.3 Matching Requests to Server Transactions has a protocol =
change (or is that not called a protocol change?).
=20
Regards,
Hisham=20

-----Original Message-----
From: ext Rosen, Brian [mailto:Brian.Rosen@marconi.com]
Sent: Tuesday, July 16, 2002 10:44 AM
To: 'ABDUL SATTAR SHAIK'; 'sip@ietf.org'
Subject: RE: [Sip] (no subject)


There were a number of relatively small things fixed.=20
For example, several ABNF errors were reported after bis-09
and before RFC3261 which were fixed.   I believe we will
provide a diff to highlight the changes.  This is harder than
you might think because the RFC editor completely redid the
formatting, so creating a diff is not really easy. =20
=20
There are no actual protocol changes.
=20
Brian

-----Original Message-----
From: ABDUL SATTAR SHAIK [mailto:abdulss@migv.mot.com]
Sent: Tuesday, July 16, 2002 2:06 AM
To: 'sip@ietf.org'
Subject: [Sip] (no subject)



Hi=20
what are the differences between rfc2543-bis-09 and rfc 3261?=20

I  didn't find a single difference in these two.=20

I think the bis-09 is  published as 3261 since bis-09 expires by august. =


AmI right ?=20

Thanks=20
abdul sattar=20



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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<TITLE></TITLE>

<META content=3D"MSHTML 5.50.4611.1300" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D688091208-22072002><FONT face=3DArial color=3D#0000ff =
size=3D2><FONT=20
color=3D#000000 size=3D3>Section 17.2.3 Matching Requests to Server=20
Transactions&nbsp;has a protocol change (or is that not called a =
protocol=20
change?).</FONT></FONT></SPAN></DIV>
<DIV><SPAN class=3D688091208-22072002><FONT face=3DArial color=3D#0000ff =
size=3D2><FONT=20
color=3D#000000 size=3D3></FONT></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D688091208-22072002><FONT=20
face=3DArial>Regards,</FONT></SPAN></DIV>
<DIV><SPAN class=3D688091208-22072002><FONT=20
face=3DArial>Hisham</FONT>&nbsp;</SPAN></DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> ext Rosen, Brian=20
  [mailto:Brian.Rosen@marconi.com]<BR><B>Sent:</B> Tuesday, July 16, =
2002 10:44=20
  AM<BR><B>To:</B> 'ABDUL SATTAR SHAIK'; =
'sip@ietf.org'<BR><B>Subject:</B> RE:=20
  [Sip] (no subject)<BR><BR></FONT></DIV>
  <DIV><SPAN class=3D857403607-16072002><FONT face=3DArial =
color=3D#0000ff>There were=20
  a number of relatively small things fixed.&nbsp;</FONT></SPAN></DIV>
  <DIV><SPAN class=3D857403607-16072002><FONT face=3DArial =
color=3D#0000ff>For=20
  example, several ABNF errors were reported after =
bis-09</FONT></SPAN></DIV>
  <DIV><SPAN class=3D857403607-16072002><FONT face=3DArial =
color=3D#0000ff>and before=20
  RFC3261 which were fixed.&nbsp;&nbsp; I believe we =
will</FONT></SPAN></DIV>
  <DIV><SPAN class=3D857403607-16072002><FONT face=3DArial =
color=3D#0000ff>provide a=20
  diff to highlight the changes.&nbsp; This is harder =
than</FONT></SPAN></DIV>
  <DIV><SPAN class=3D857403607-16072002><FONT face=3DArial =
color=3D#0000ff>you might=20
  think because the RFC editor completely redid the</FONT></SPAN></DIV>
  <DIV><SPAN class=3D857403607-16072002><FONT face=3DArial =
color=3D#0000ff>formatting,=20
  so creating a diff is not really easy.&nbsp; </FONT></SPAN></DIV>
  <DIV><SPAN class=3D857403607-16072002><FONT face=3DArial=20
  color=3D#0000ff></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D857403607-16072002><FONT face=3DArial =
color=3D#0000ff>There are=20
  no actual </FONT></SPAN><SPAN class=3D857403607-16072002><FONT =
face=3DArial=20
  color=3D#0000ff>protocol changes</FONT></SPAN><SPAN=20
  class=3D857403607-16072002><FONT face=3DArial =
color=3D#0000ff>.</FONT></SPAN></DIV>
  <DIV><SPAN class=3D857403607-16072002><FONT face=3DArial=20
  color=3D#0000ff></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D857403607-16072002><FONT face=3DArial=20
  color=3D#0000ff>Brian</FONT></SPAN></DIV>
  <BLOCKQUOTE=20
  style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid">
    <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
    size=3D2>-----Original Message-----<BR><B>From:</B> ABDUL SATTAR =
SHAIK=20
    [mailto:abdulss@migv.mot.com]<BR><B>Sent:</B> Tuesday, July 16, 2002 =
2:06=20
    AM<BR><B>To:</B> 'sip@ietf.org'<BR><B>Subject:</B> [Sip] (no=20
    subject)<BR><BR></FONT></DIV>
    <P><FONT face=3DArial size=3D2>Hi</FONT> <BR><FONT face=3DArial =
size=3D2>what are=20
    the differences between rfc2543-bis-09 and rfc 3261?</FONT> </P>
    <P><FONT face=3DArial size=3D2>I&nbsp; didn't find a single =
difference in these=20
    two.</FONT> </P>
    <P><FONT face=3DArial size=3D2>I think the bis-09 is&nbsp; published =
as 3261=20
    since bis-09 expires by august.</FONT> </P>
    <P><FONT face=3DArial size=3D2>AmI right ?</FONT> </P>
    <P><FONT face=3DArial size=3D2>Thanks</FONT> <BR><FONT face=3DArial =
size=3D2>abdul=20
    sattar</FONT> </P><BR></BLOCKQUOTE></BLOCKQUOTE></BODY></HTML>

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From daemon@optimus.ietf.org  Mon Jul 22 09:54:54 2002
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From: Adam Roach <adam@dynamicsoft.com>
To: sip@ietf.org
Subject: RE: [Sip] Proposed change in draft-ietf-sip-nat
Date: Mon, 22 Jul 2002 08:26:02 -0500
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I would like to apologise to Pete for the comments below.
They were intended as banter between colleagues and not
meant to be cc'ed to the list.

/a

> -----Original Message-----
> From: Adam Roach [mailto:adam@dynamicsoft.com]
> Sent: Sunday, July 14, 2002 20:15
> To: Pete Cordell; Jonathan Rosenberg; Cullen Jennings
> Cc: sip@ietf.org
> Subject: RE: [Sip] Proposed change in draft-ietf-sip-nat
> 
> 
> > -----Original Message-----
> > From: Pete Cordell [mailto:pcordell@ridgewaysystems.com]
> >
> > So why not acknowledge the editorial error (which
> > requires no action) and move on.
> 
> Wow. I never thought I'd see the day that the SIP Working Group
> mailing list devolved to the level of an old-fashioned Usenet-style
> spelling flame.
> 
> Thanks, Pete. It's a first.
> 
> /a
> 
> _______________________________________________
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> 

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From daemon@optimus.ietf.org  Mon Jul 22 10:03:35 2002
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Subject: [Sip] Re: [Simple] -05 MESSAGE and Expires header
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It seems to me that if the UAC cares about intermediaries seeing the 
expires header, it would put an expires header in the outer message.

Vijay K. Gurbani wrote:
> This has been an interesting discussion with more people weighing
> in the 'leave it alone' category.  However, here is an additional
> point to consider: Expires is one of the headers that can be
> encrypted.  An S/MIME encrypted MESSAGE request containing 'Expires: 0'
> arriving at a relay UAS will get the same processing as will
> an unencrypted MESSAGE with 'Expires: 3600' -- probably not good.  The
> Priority header, on the other hand, is not permitted to be encrypted
> (Table 3, RFC 3261); thus relays can make use of it.
> 
> Jonathan Rosenberg gave a concise explanation on why Priority and
> 'Expires: 0' to mean immediate delivery are different:
> 
>  > I believe that priority is quite different. A message can be for
>  > immediate delivery, and be either urgent ("my house is on fire") or
>  > not ("leaving now for lunch"). A message can be OK for storage (some
>  > non-zero expires) and be urgent ("I need that business presentation
>  > by 10pm!") or non-urgent ("Did you see that movie"). The fact that two
>  > things can be used orthogonally is a sign that they are, in fact, two
>  > different things.
> 
> In this context, maybe Priority is not the right header, but it is
> the closest we have in base SIP to impart immediate delivery.
> 
> The whole point of 'Expires: 0' in the MESSAGE I-D is for the relay UAS
> and the destination UAS to act on this request immediately.  Since the
> Expire header can be encrypted, the directive of immediate delivery is
> lost at the relay UAS.  Priority preserves it, and in fact, to avoid any
> further ambiguity,  the MESSAGE I-D can further restrict the presence of
> the Priority header in only those IM messages that denote urgency of
> some sort, and thus need special processing by the relays.
> 
> Regards,
> 
> - vijay




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From daemon@optimus.ietf.org  Mon Jul 22 10:52:00 2002
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Ben Campbell wrote:
> It seems to me that if the UAC cares about intermediaries seeing the 
> expires header, it would put an expires header in the outer message.

It could; but really, why duplicate the information?

RFC 2076 (Common Internet Message Headers) says that 'Priority' can
"influence transmission speed and delivery".  Which seems to fit exactly
with what we want to do.

In fact, the more I think of it, the more I am convinced that
overloading 'Expires: 0' to mean something completely different in
MESSAGE is not a good idea; especially after a precedent has been
established by REGISTER and SUBSCRIBE to use 'Expires: 0' to mean
'no longer valid'.

It's been pointed out that Priority is not the right header, but I just
don't see why not.  Expires is deemed to be an end-to-end header; as
such, why use it for routing purposes by relays?  Whereas, Priority
appears to be exactly for this purpose.

Adam has proposed the 'Urgency' header; but I don't think we need
to make up yet another new header for this and update deployed parsers;
Priority handles many shades of urgencies and emergencies.

IMHO, giving 'Expires: 0' an overloaded meaning, while having a header
('Priority') that can be used for expedited delivery is what leads to
problems of the kind Jonathan alluded to:

 > A message can be for immediate delivery, and be either urgent ("my
 > house is on fire") or not ("leaving now for lunch"). A message can be
 > OK for storage (some non-zero expires) and be urgent ("I need that
 > business presentation by 10pm!") or non-urgent ("Did you see that
 > movie").

If, on the other hand, we simply specify 'Priority: Urgent' (or
'Priority: Emergency'), both the UA and the relays should act
immediately on that message.  No need to add new semantics to 'Expires:
0'.

In the final analysis, what is the harm if we can make the MESSAGE PDU
self explanatory?  I would think that almost everyone will agree that
'Priority: Urgent' is more self-documenting than 'Expires: 0'.  As I
suggested earlier, the MESSAGE I-D can mandate that 'Priority: Urgent'
or 'Priority: Emergency' be only used when, as 3261 says, "life, limb,
or property are in imminent danger."

Regards,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and Services
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216


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From daemon@optimus.ietf.org  Mon Jul 22 11:37:37 2002
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        Ben Campbell
	 <bcampbell@dynamicsoft.com>
Cc: SIP LIST <sip@ietf.org>
Subject: RE: [Sip] Re: [Simple] -05 MESSAGE and Expires header
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> -----Original Message-----
> From: Vijay K. Gurbani [mailto:vkg@lucent.com]
> 
> Adam has proposed the 'Urgency' header

No, I didn't.

I attempted to outline that what we're talking about isn't
priority in any sense of the word.

We're talking about urgency.

I asked Hisham whether he was making a proposal for a new
header -- but it was asked with a certain amount of irony.
I didn't expect that he was doing so, and crafted the rest
of my message with the assumption that his answer would be
"no".

My argument remains that "Expires" is the most appropriate
header for this purpose.

/a

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From daemon@optimus.ietf.org  Mon Jul 22 12:04:05 2002
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From: "Vijay K. Gurbani" <vkg@lucent.com>
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CC: Ben Campbell <bcampbell@dynamicsoft.com>, SIP LIST <sip@ietf.org>
Subject: Re: [Sip] Re: [Simple] -05 MESSAGE and Expires header
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Adam Roach wrote:
>>-----Original Message-----
>>From: Vijay K. Gurbani [mailto:vkg@lucent.com]
>>
>>Adam has proposed the 'Urgency' header
> 
> No, I didn't.

Okay; my apologies.  I did detect the irony, but absent a face-
to-face conversation, I wasn't sure if you were being serious or
flippant.  Glad to know we won't be making up a new header.

> I attempted to outline that what we're talking about isn't
> priority in any sense of the word.
> 
> We're talking about urgency.

If I may, I think we are imparting urgency to an IM based on the
*content* of the IM.  In other words, to use one of your examples:

> High Priority, Low Urgency:
>   Note to new residents: failure to file income taxes by April
>   15th can lead to substantial penalties.

Absence other information (like the date that this IM was sent),
urgency is meaningless.  For example, if this IM was sent on April
14th, I would think that it would be classified with high urgency.
If, OTOH, it was sent on April 16th, its urgency is extremely low.
And a relay should not be put in position to use many heuristics
for figuring out which IMs should get immediate attention.

> My argument remains that "Expires" is the most appropriate
> header for this purpose.

Right; I see I haven't convinced you yet :-)

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and Services
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216


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From daemon@optimus.ietf.org  Mon Jul 22 12:56:04 2002
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CC: Adam Roach <adam@dynamicsoft.com>, SIP LIST <sip@ietf.org>
Subject: Re: [Sip] Re: [Simple] -05 MESSAGE and Expires header
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Vijay K. Gurbani wrote:
> Adam Roach wrote:
> 
>>> -----Original Message-----
>>> From: Vijay K. Gurbani [mailto:vkg@lucent.com]
>>>
>>> Adam has proposed the 'Urgency' header
>>
>>
>> No, I didn't.
> 
> 
> Okay; my apologies.  I did detect the irony, but absent a face-
> to-face conversation, I wasn't sure if you were being serious or
> flippant.  Glad to know we won't be making up a new header.
> 
>> I attempted to outline that what we're talking about isn't
>> priority in any sense of the word.
>>
>> We're talking about urgency.
> 
> 
> If I may, I think we are imparting urgency to an IM based on the
> *content* of the IM.  In other words, to use one of your examples:
> 
>> High Priority, Low Urgency:
>>   Note to new residents: failure to file income taxes by April
>>   15th can lead to substantial penalties.
> 
> 
> Absence other information (like the date that this IM was sent),
> urgency is meaningless.  For example, if this IM was sent on April
> 14th, I would think that it would be classified with high urgency.
> If, OTOH, it was sent on April 16th, its urgency is extremely low.
> And a relay should not be put in position to use many heuristics
> for figuring out which IMs should get immediate attention.
> 

OK, let me extend Adam's example: The message is sent on March 15. It 
has high priority, but low urgency. (And I assume that Adam was not 
suggesting that the relay understand the content of the note to 
determine urgency and priority--he was simply suggesting a situation 
where a sender might mark the message as high priority but low urgency.)


>> My argument remains that "Expires" is the most appropriate
>> header for this purpose.
> 
> 
> Right; I see I haven't convinced you yet :-)
> 
> - vijay




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Date: Mon, 22 Jul 2002 10:06:45 -0700 (PDT)
From: Sean Olson <seancolson@yahoo.com>
Subject: Re: [Sip] Re: [Simple] -05 MESSAGE and Expires header
To: Ben Campbell <bcampbell@dynamicsoft.com>,
        "Vijay K. Gurbani" <vkg@lucent.com>
Cc: Adam Roach <adam@dynamicsoft.com>, SIP LIST <sip@ietf.org>
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So we have three things we might want to 
represent:

1) Priority -- we use the Priority header for this
2) Timeliness of content -- we use Expires: header
   for this
3) Urgency -- we don't currently have a way to 
   represent this

Obviously 2 & 3 are related but not identical.
The message doesn't have to be urgent in order
for you to request immediate delivery. I think
solving (2) is enough for now. It is the more
interesting case for gateways and other
intermediaries that have to decide between a 
store-and-forward or immediate delivery approach.
In other words, stick with what we have today in
the draft.

/sean

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From daemon@optimus.ietf.org  Mon Jul 22 14:31:09 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
To: <sip@ietf.org>
Cc: <mankin@isi.edu>
Date: Mon, 22 Jul 2002 13:04:26 -0500
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Subject: [Sip] Response Summary -- Should service route discovery be standards track or p-header?
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I raised the follwing on July 9:
> Should the SIP Service route Discovery draft be an 
> informational P-header or should it be standards track? Some 
> have argued that it is more general than usual for P-headers. 
> It seems to me to be useful whenever it is useful for a 
> registrar to return routing infromation that UAs may use when 
> they want request routing services from the domain served by 
> that registrar.

Almost all response were in favor of making this standards track.

Several people dissented with arguments along the lines of "Let's make
it a P-header so that we can use this as a test case for proving that
P-headers might not be a good idea". While I'm not entirely opposed to
this sort of protest action, it doesn't seem right to me in this case.

So, I think we have a rough consensus for taking this work onto
standards track.

I believe very little change is needed to the existing document to adapt
it for standards-track -- basically, delete the P-, and update the
references to include 3261. The draft has already had extensive review
within the working group, so I think we should be able to complete a new
WGLC fairly easily. I would suggest that we adopt it as a charter item
with a milestone of delivery to IESG in August, with expected delivery
being "very early August".

Comments?

--
Dean



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From daemon@ns.ietf.org  Mon Jul 22 17:34:20 2002
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From: "Taja Ja" <hayanitm@hotmail.com>
To: sip@ietf.org
Cc: mkadoch@ele.etsmtl.ca
Date: Thu, 18 Jul 2002 13:51:27 -0400
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Subject: [Sip] SIP and RTP
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Hi ,

  If someone can help understand how SIP uses RTP.
  (we have two users agent installed on 2 different PCs
   they SIP together but I don't know how to call/use RTP)

  Any help is greatly apperciated.

Thanks for your time.Taher.




_________________________________________________________________
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http://photos.msn.com/support/worldwide.aspx



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From daemon@ns.ietf.org  Mon Jul 22 17:56:56 2002
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Date: Mon, 22 Jul 2002 16:30:31 -0500
To: "Taja Ja" <hayanitm@hotmail.com>, sip@ietf.org
From: "James M. Polk" <jmpolk@cisco.com>
Subject: Re: [Sip] SIP and RTP
Cc: mkadoch@ele.etsmtl.ca
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At 01:51 PM 7/18/2002 -0400, Taja Ja wrote:
>Hi ,
>
>  If someone can help understand how SIP uses RTP.
>  (we have two users agent installed on 2 different PCs
>   they SIP together but I don't know how to call/use RTP)

Within the INVITE is SDP (capabilities negotiation) for the type of (for
example) RTP session the UAC wants to have with the UAS. The SDP specifies
the payload type/codec and other such details for that proposed session.
Simplistically, if the UAS replies with a 200 OK Response, the session is
created with the SDP parameters specified from the UAC (after the UAC sends
a final ACK to the UAS)

RFC 3264 is a good reference for this explanation in the Offer-Answer Model
where both UAs become aware of the capabilities of each other prior to
session establishment.

>
>  Any help is greatly apperciated.
>
>Thanks for your time.Taher.
>
>
>
>
>_________________________________________________________________
>MSN Photos is the easiest way to share and print your photos: 
>http://photos.msn.com/support/worldwide.aspx
>
>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip


cheers,
James 

              *************************************
"People generally demand more respect for their own rights than 
                         they are willing to allow for others"



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From daemon@ns.ietf.org  Tue Jul 23 01:36:53 2002
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Date: Tue, 23 Jul 2002 13:55:37 +0900
From: Shingo Fujimoto <shingo_fujimoto@jp.fujitsu.com>
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Subject: [Sip] language info in SIP headers
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Hello. I would like to raise the new issue to solve.

SIP RFC specified UTF-8 friendly transport, but I think that did not
specified the way to use multi bytes characters in header part of
SIP messages.

If I use Japanese Kanji letters in From: header (as display name),
the receiver UA cannot determine which language of characters are
used in UTF-8 encoded text. Additionally, since UTF-8 CJK kanji
characters are overlapped between Chinese, Japanese, and Korean,
sometimes UA can not determine which font should be used to display
them.

MIME part 3 and updated RFC2231 introduced some solution for this
problem with combination of character set and language tag. However
RFC2231 always required escaping 8bit-flaged characters, there are
huge overhead to express multi byte characters.

As another option, recent Unicode 3.x specification includes
"language tag" characters to solve this problem, but it is not
common to support such new feature in the any UTF-8 handling
libraries. I guess that results corrupted display in UAs.

My suggestion about this, we should prepare supplemental document
for SIP which specify the way to express international characters
with language information in header part (modified RFC2231 or UTF-8
with "lang tag" support).

I would like to hear your opinion.

Thank you.
-- 
======================================
         SHINGO FUJIMOTO
    FUJITSU LABORATORIES LIMITED
E-mail: shingo_fujimoto@jp.fujitsu.com



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From daemon@ns.ietf.org  Tue Jul 23 03:01:15 2002
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Subject: RE: [Sip] Re: [Simple] -05 MESSAGE and Expires header
Date: Tue, 23 Jul 2002 09:28:50 +0300
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Thread-Topic: [Sip] Re: [Simple] -05 MESSAGE and Expires header
Thread-Index: AcIxokD1X/EzOSp1Q4a/HTta87/C2QAbiMkg
To: <seancolson@yahoo.com>, <bcampbell@dynamicsoft.com>, <vkg@lucent.com>
Cc: <adam@dynamicsoft.com>, <sip@ietf.org>
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So, a MESSAGE with Expires: 10 means it has 10 seconds before it's stale while a MESSAGE with Expires: 0 means it will never be stale?

How do u inform the receiver that a message is stale?

/Hisham

> -----Original Message-----
> From: ext Sean Olson [mailto:seancolson@yahoo.com]
> Sent: Monday, July 22, 2002 8:07 PM
> To: Ben Campbell; Vijay K. Gurbani
> Cc: Adam Roach; SIP LIST
> Subject: Re: [Sip] Re: [Simple] -05 MESSAGE and Expires header
> 
> 
> So we have three things we might want to 
> represent:
> 
> 1) Priority -- we use the Priority header for this
> 2) Timeliness of content -- we use Expires: header
>    for this
> 3) Urgency -- we don't currently have a way to 
>    represent this
> 
> Obviously 2 & 3 are related but not identical.
> The message doesn't have to be urgent in order
> for you to request immediate delivery. I think
> solving (2) is enough for now. It is the more
> interesting case for gateways and other
> intermediaries that have to decide between a 
> store-and-forward or immediate delivery approach.
> In other words, stick with what we have today in
> the draft.
> 
> /sean
> 
> __________________________________________________
> Do You Yahoo!?
> Yahoo! Health - Feel better, live better
> http://health.yahoo.com
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

_______________________________________________
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Date: Mon, 22 Jul 2002 23:39:06 -0700 (PDT)
From: Sean Olson <seancolson@yahoo.com>
Subject: RE: [Sip] Re: [Simple] -05 MESSAGE and Expires header
To: hisham.khartabil@nokia.com, bcampbell@dynamicsoft.com, vkg@lucent.com
Cc: adam@dynamicsoft.com, sip@ietf.org
In-Reply-To: <2038BCC78B1AD641891A0D1AE133DBB7C203F2@esebe019.NOE.Nokia.com>
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I fail to see why Expires: 0 means the message
will never expire. How does 0 equate to infinity?

An Expires: 0 from the sender, in my opinion,
should indicate that this message should be delivered
immediately or not at all. I don't think it should
be used for the sender (or gateway or proxy) to
indicate that a message is "stale".

By stale,
I assume you mean a message that was intended for
immediate delivery, but was actually stored instead
-- this is not something existing SIP headers do a
good job of representing. I wouldn't try to hack
Expires: to fit this. Furthermore, I don't think
this is a problem that the message draft should have
to solve. Until someone comes up with a more complete
draft for offline message retrieval, I think we can
punt this. Or is that what you were offering to do?


--- hisham.khartabil@nokia.com wrote:
> So, a MESSAGE with Expires: 10 means it has 10
> seconds before it's stale while a MESSAGE with
> Expires: 0 means it will never be stale?
> 
> How do u inform the receiver that a message is
> stale?
> 
> /Hisham
> 
> > -----Original Message-----
> > From: ext Sean Olson [mailto:seancolson@yahoo.com]
> > Sent: Monday, July 22, 2002 8:07 PM
> > To: Ben Campbell; Vijay K. Gurbani
> > Cc: Adam Roach; SIP LIST
> > Subject: Re: [Sip] Re: [Simple] -05 MESSAGE and
> Expires header
> > 
> > 
> > So we have three things we might want to 
> > represent:
> > 
> > 1) Priority -- we use the Priority header for this
> > 2) Timeliness of content -- we use Expires: header
> >    for this
> > 3) Urgency -- we don't currently have a way to 
> >    represent this
> > 
> > Obviously 2 & 3 are related but not identical.
> > The message doesn't have to be urgent in order
> > for you to request immediate delivery. I think
> > solving (2) is enough for now. It is the more
> > interesting case for gateways and other
> > intermediaries that have to decide between a 
> > store-and-forward or immediate delivery approach.
> > In other words, stick with what we have today in
> > the draft.
> > 
> > /sean
> > 
> > __________________________________________________
> > Do You Yahoo!?
> > Yahoo! Health - Feel better, live better
> > http://health.yahoo.com
> > 
> > _______________________________________________
> > Sip mailing list 
> https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP
> Protocol
> > Use sip-implementors@cs.columbia.edu for questions
> on current sip
> > Use sipping@ietf.org for new developments on the
> application of sip
> > 


__________________________________________________
Do You Yahoo!?
Yahoo! Health - Feel better, live better
http://health.yahoo.com

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@ns.ietf.org  Tue Jul 23 03:03:24 2002
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From: Sean Olson <seancolson@yahoo.com>
Subject: RE: [Sip] Re: [Simple] -05 MESSAGE and Expires header
To: hisham.khartabil@nokia.com, bcampbell@dynamicsoft.com, vkg@lucent.com
Cc: adam@dynamicsoft.com, sip@ietf.org
In-Reply-To: <2038BCC78B1AD641891A0D1AE133DBB7C203F2@esebe019.NOE.Nokia.com>
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I fail to see why Expires: 0 means the message
will never expire. How does 0 equate to infinity?

An Expires: 0 from the sender, in my opinion,
should indicate that this message should be delivered
immediately or not at all. I don't think it should
be used for the sender (or gateway or proxy) to
indicate that a message is "stale".

By stale,
I assume you mean a message that was intended for
immediate delivery, but was actually stored instead
-- this is not something existing SIP headers do a
good job of representing. I wouldn't try to hack
Expires: to fit this. Furthermore, I don't think
this is a problem that the message draft should have
to solve. Until someone comes up with a more complete
draft for offline message retrieval, I think we can
punt this. Or is that what you were offering to do?


--- hisham.khartabil@nokia.com wrote:
> So, a MESSAGE with Expires: 10 means it has 10
> seconds before it's stale while a MESSAGE with
> Expires: 0 means it will never be stale?
> 
> How do u inform the receiver that a message is
> stale?
> 
> /Hisham
> 
> > -----Original Message-----
> > From: ext Sean Olson [mailto:seancolson@yahoo.com]
> > Sent: Monday, July 22, 2002 8:07 PM
> > To: Ben Campbell; Vijay K. Gurbani
> > Cc: Adam Roach; SIP LIST
> > Subject: Re: [Sip] Re: [Simple] -05 MESSAGE and
> Expires header
> > 
> > 
> > So we have three things we might want to 
> > represent:
> > 
> > 1) Priority -- we use the Priority header for this
> > 2) Timeliness of content -- we use Expires: header
> >    for this
> > 3) Urgency -- we don't currently have a way to 
> >    represent this
> > 
> > Obviously 2 & 3 are related but not identical.
> > The message doesn't have to be urgent in order
> > for you to request immediate delivery. I think
> > solving (2) is enough for now. It is the more
> > interesting case for gateways and other
> > intermediaries that have to decide between a 
> > store-and-forward or immediate delivery approach.
> > In other words, stick with what we have today in
> > the draft.
> > 
> > /sean
> > 
> > __________________________________________________
> > Do You Yahoo!?
> > Yahoo! Health - Feel better, live better
> > http://health.yahoo.com
> > 
> > _______________________________________________
> > Sip mailing list 
> https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP
> Protocol
> > Use sip-implementors@cs.columbia.edu for questions
> on current sip
> > Use sipping@ietf.org for new developments on the
> application of sip
> > 


__________________________________________________
Do You Yahoo!?
Yahoo! Health - Feel better, live better
http://health.yahoo.com

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Jul 23 04:23:39 2002
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From: Henning Schulzrinne <hgs@cs.columbia.edu>
Organization: Columbia University
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To: Shingo Fujimoto <shingo_fujimoto@jp.fujitsu.com>
CC: SIP-WG <sip@ietf.org>
Subject: Re: [Sip] language info in SIP headers
References: <3D3CE1C9.4080505@jp.fujitsu.com>
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This is actually a fairly thorny problem. From a previous private
discussion, there are only a few possible solutions:

- RFC 2482 or http://www.unicode.org/unicode/reports/tr27/#tag provide
language tagging, as you noted. While I have not seen an official
statement from the IESG on this, I believe 10646 language tagging has
generally been deprecated. It's not quite clear whether the reasons (I
believe, nesting) for that apply to short text such as Subject or
Organization.

- For header fields that allow parameters, I believe the best option is
to add a language parameter, as in

From: "Somebody" <sip:somebody@somewhere.com> ;language="en"

This doesn't allow multiple languages within the same header field, but
is probably useful and sufficient as a hint for text-to-speech and other
UI rendering. It is also backward-compatible. This works for From, To
and a few others, such as the NAI header fields.

- For header fields that don't allow parameters (Subject, Organization,
etc.), there are no good options. In general, I believe we should avoid
such header fields in the future. One possible kludge suggested in
private conversation would be to add a descriptive header that tells you
what language each header field contains. Or, simpler but less
space-efficient, something like

Subject-Language: jp
Organization-Language: en-us

Any other suggestions?

Shingo Fujimoto wrote:
> Hello. I would like to raise the new issue to solve.
> 
> SIP RFC specified UTF-8 friendly transport, but I think that did not
> specified the way to use multi bytes characters in header part of
> SIP messages.
> 
> If I use Japanese Kanji letters in From: header (as display name),
> the receiver UA cannot determine which language of characters are
> used in UTF-8 encoded text. Additionally, since UTF-8 CJK kanji
> characters are overlapped between Chinese, Japanese, and Korean,
> sometimes UA can not determine which font should be used to display
> them.
> 
> MIME part 3 and updated RFC2231 introduced some solution for this
> problem with combination of character set and language tag. However
> RFC2231 always required escaping 8bit-flaged characters, there are
> huge overhead to express multi byte characters.
> 
> As another option, recent Unicode 3.x specification includes
> "language tag" characters to solve this problem, but it is not
> common to support such new feature in the any UTF-8 handling
> libraries. I guess that results corrupted display in UAs.
> 
> My suggestion about this, we should prepare supplemental document
> for SIP which specify the way to express international characters
> with language information in header part (modified RFC2231 or UTF-8
> with "lang tag" support).
> 
> I would like to hear your opinion.
> 
> Thank you.



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Jul 23 04:26:47 2002
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From: hisham.khartabil@nokia.com
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Subject: RE: [Sip] Re: [Simple] -05 MESSAGE and Expires header
Date: Tue, 23 Jul 2002 10:51:24 +0300
Message-ID: <2038BCC78B1AD641891A0D1AE133DBB7C203F4@esebe019.NOE.Nokia.com>
Thread-Topic: [Sip] Re: [Simple] -05 MESSAGE and Expires header
Thread-Index: AcIyE6a+Pb+8QXmiTme4Glu1d52crAACMXag
To: <seancolson@yahoo.com>, <bcampbell@dynamicsoft.com>, <vkg@lucent.com>
Cc: <adam@dynamicsoft.com>, <sip@ietf.org>
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I would like some interpretation for a few messages in terms of immediate delivery, urgency and expiration of a message:

1.
MESSAGE ...
Priority: non-urgent
Expires: 0
...
I'm leaving for lunch right now

2.
MESSAGE ...
Priority: emergency
Expires: 3600
...
your house is on fire

3.
MESSAGE ...
...
Date: Tue, 23 Jul 2002 7:25:00 GMT
Expires: 60

4.
MESSAGE ...
...
Date: Tue, 23 Jul 2002 7:25:00 GMT
Expires: 0


Do messages 1 and 2 get immediate delivery?

Another question: This immediate delivery debate is only talking about Store and Forward type of servers of gateways. What about other intermediaries? They certainly can't use the expires-header to decide on speed of delivery of a message. We need a new mechanism here. Again I ask the question: Is this done in IEPREP?

Regards,
Hisham

> -----Original Message-----
> From: ext Sean Olson [mailto:seancolson@yahoo.com]
> Sent: Tuesday, July 23, 2002 9:39 AM
> To: Khartabil Hisham (NMP/Helsinki); bcampbell@dynamicsoft.com;
> vkg@lucent.com
> Cc: adam@dynamicsoft.com; sip@ietf.org
> Subject: RE: [Sip] Re: [Simple] -05 MESSAGE and Expires header
> 
> 
> I fail to see why Expires: 0 means the message
> will never expire. How does 0 equate to infinity?
> 
> An Expires: 0 from the sender, in my opinion,
> should indicate that this message should be delivered
> immediately or not at all. I don't think it should
> be used for the sender (or gateway or proxy) to
> indicate that a message is "stale".
> 
> By stale,
> I assume you mean a message that was intended for
> immediate delivery, but was actually stored instead
> -- this is not something existing SIP headers do a
> good job of representing. I wouldn't try to hack
> Expires: to fit this. Furthermore, I don't think
> this is a problem that the message draft should have
> to solve. Until someone comes up with a more complete
> draft for offline message retrieval, I think we can
> punt this. Or is that what you were offering to do?
> 
> 
> --- hisham.khartabil@nokia.com wrote:
> > So, a MESSAGE with Expires: 10 means it has 10
> > seconds before it's stale while a MESSAGE with
> > Expires: 0 means it will never be stale?
> > 
> > How do u inform the receiver that a message is
> > stale?
> > 
> > /Hisham
> > 
> > > -----Original Message-----
> > > From: ext Sean Olson [mailto:seancolson@yahoo.com]
> > > Sent: Monday, July 22, 2002 8:07 PM
> > > To: Ben Campbell; Vijay K. Gurbani
> > > Cc: Adam Roach; SIP LIST
> > > Subject: Re: [Sip] Re: [Simple] -05 MESSAGE and
> > Expires header
> > > 
> > > 
> > > So we have three things we might want to 
> > > represent:
> > > 
> > > 1) Priority -- we use the Priority header for this
> > > 2) Timeliness of content -- we use Expires: header
> > >    for this
> > > 3) Urgency -- we don't currently have a way to 
> > >    represent this
> > > 
> > > Obviously 2 & 3 are related but not identical.
> > > The message doesn't have to be urgent in order
> > > for you to request immediate delivery. I think
> > > solving (2) is enough for now. It is the more
> > > interesting case for gateways and other
> > > intermediaries that have to decide between a 
> > > store-and-forward or immediate delivery approach.
> > > In other words, stick with what we have today in
> > > the draft.
> > > 
> > > /sean
> > > 
> > > __________________________________________________
> > > Do You Yahoo!?
> > > Yahoo! Health - Feel better, live better
> > > http://health.yahoo.com
> > > 
> > > _______________________________________________
> > > Sip mailing list 
> > https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP
> > Protocol
> > > Use sip-implementors@cs.columbia.edu for questions
> > on current sip
> > > Use sipping@ietf.org for new developments on the
> > application of sip
> > > 
> 
> 
> __________________________________________________
> Do You Yahoo!?
> Yahoo! Health - Feel better, live better
> http://health.yahoo.com
> 

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Jul 23 06:15:42 2002
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Date: Tue, 23 Jul 2002 05:48:59 -0400
From: Henning Schulzrinne <hgs@cs.columbia.edu>
Organization: Columbia University
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To: Shingo Fujimoto <shingo_fujimoto@jp.fujitsu.com>
CC: SIP-WG <sip@ietf.org>
Subject: Re: [Sip] language info in SIP headers
References: <3D3CE1C9.4080505@jp.fujitsu.com> <3D3D0C1A.7020307@cs.columbia.edu> <3D3D22C6.2080208@jp.fujitsu.com>
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> The reason I prefer this approach is that SIP messages must be small
> if it runs over UDP. "language tag" will add only few bytes.
> 
> Problem is backward compatibility with widely deployed SIP implementations.
> Another option is use MIME part3 with small modification.
> Since current MIME part3 did not allow to use any 8 bit flagged
> characters in MIME headers, encoding always escapes UTF-8 characters.
> 
> This is an example of MIME part3 header.
> 
> From: =?UTF-8*JP?B?<base64 encoded utf-8 string>?= <sip:foo@bar.org>
> 
> "base64" part is escaped and increased its size with 1.5 times or more.
> 
> If we define new encoding, say 'U', which only escape special
> charactors ('=','?',and '%' with '%xx' escaping).
> 
> From: =?UTF-8*JP?U?<raw utf-8 string>?= <sip:foo@bar.org>
> 
> This extension saves overhead on 8 bit charactors, and will not
> suffer any older implementations.
> 
> Additionally, if we can assume utf-8 and 'U' encoding, then we can
> minimize the escaping syntax too.
> 
> From: =?JP?<raw utf-8 string>?= <sip:foo@bar.org>
> 
> This still valid as display name value.
> 
> Both Keith and Ned suggested me that if we define 'U' encoding, it
> should be limited within SIP specification.
> 

I would suggest you write up a short I-D on the subject.



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From daemon@optimus.ietf.org  Tue Jul 23 06:15:45 2002
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To: Henning Schulzrinne <hgs@cs.columbia.edu>
CC: SIP-WG <sip@ietf.org>
Subject: Re: [Sip] language info in SIP headers
References: <3D3CE1C9.4080505@jp.fujitsu.com> <3D3D0C1A.7020307@cs.columbia.edu>
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I am happy to hear you already had some discussions.

Henning Schulzrinne wrote:

> - RFC 2482 or http://www.unicode.org/unicode/reports/tr27/#tag provide
> language tagging, as you noted. While I have not seen an official
> statement from the IESG on this, I believe 10646 language tagging has
> generally been deprecated. It's not quite clear whether the reasons (I
> believe, nesting) for that apply to short text such as Subject or
> Organization.


You are probably right.

Here is the comment from Ned Freed, one of the RFC2231 authors:

> Language indicators in a pure UTF-8 environment like SIP headers have to be
> stored as part of the UTF-8 string itself. The plane 14 language tags described
> in
> 
>    http://www.unicode.org/unicode/reports/tr27/#tag
> 
> are the only means of doing this that I'm aware of. However, the use
> of such tags is, as the text indicates, strongly discouraged.
> 
> It is the Unicode Technical Committee's belief that
> 
>   The requirement for language information embedded in plain text data is often
>   overstated.
> 
> As such, I see little chance for support for such tagging in the context of SIP
> headers being widely deployed.
> 
> 				Ned
> 

The reason I prefer this approach is that SIP messages must be small
if it runs over UDP. "language tag" will add only few bytes.

Problem is backward compatibility with widely deployed SIP implementations.


> - For header fields that allow parameters, I believe the best option is
> to add a language parameter, as in
> 
> From: "Somebody" <sip:somebody@somewhere.com> ;language="en"
> 
> This doesn't allow multiple languages within the same header field, but
> is probably useful and sufficient as a hint for text-to-speech and other
> UI rendering. It is also backward-compatible. This works for From, To
> and a few others, such as the NAI header fields.


I think this approach is fair. I guess it is quite rare to use
multiple languages are used in short text field.


> - For header fields that don't allow parameters (Subject, Organization,
> etc.), there are no good options. In general, I believe we should avoid
> such header fields in the future. One possible kludge suggested in
> private conversation would be to add a descriptive header that tells you
> what language each header field contains. Or, simpler but less
> space-efficient, something like
> 
> Subject-Language: jp
> Organization-Language: en-us


I think these are also as good as previous one.

> Any other suggestions?


Another option is use MIME part3 with small modification.
Since current MIME part3 did not allow to use any 8 bit flagged
characters in MIME headers, encoding always escapes UTF-8 characters.

This is an example of MIME part3 header.

From: =?UTF-8*JP?B?<base64 encoded utf-8 string>?= <sip:foo@bar.org>

"base64" part is escaped and increased its size with 1.5 times or more.

If we define new encoding, say 'U', which only escape special
charactors ('=','?',and '%' with '%xx' escaping).

From: =?UTF-8*JP?U?<raw utf-8 string>?= <sip:foo@bar.org>

This extension saves overhead on 8 bit charactors, and will not
suffer any older implementations.

Additionally, if we can assume utf-8 and 'U' encoding, then we can
minimize the escaping syntax too.

From: =?JP?<raw utf-8 string>?= <sip:foo@bar.org>

This still valid as display name value.

Both Keith and Ned suggested me that if we define 'U' encoding, it
should be limited within SIP specification.

-- 
======================================
         SHINGO FUJIMOTO
    FUJITSU LABORATORIES LIMITED
E-mail: shingo_fujimoto@jp.fujitsu.com



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From daemon@optimus.ietf.org  Tue Jul 23 12:20:36 2002
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CC: hisham.khartabil@nokia.com, bcampbell@dynamicsoft.com,
        adam@dynamicsoft.com, sip@ietf.org
Subject: Re: [Sip] Re: [Simple] -05 MESSAGE and Expires header
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Sean Olson wrote:
> I fail to see why Expires: 0 means the message
> will never expire. How does 0 equate to infinity?
> 
> An Expires: 0 from the sender, in my opinion,
> should indicate that this message should be delivered
> immediately or not at all. I don't think it should
> be used for the sender (or gateway or proxy) to
> indicate that a message is "stale".

Staleness is what 'Expires: 0' currently implies for REGISTER
and SUBSCRIBE.  And I realize that I am fighting a loosing battle,
but for all it is worth, it will be good to accord only that semantic
to 'Expires: 0', and no more.

Now, before everyone flames me, I do realize that the MESSAGE I-D does
not allow for this behavior, nor are there any requirements in
section 5.4 of RFC 2779 to that effect.  But there have been many
times when I have sent an email which I wish I could retract.  If
in the future we want to do the same for stored IMs (i.e. ones that
elicited a 202 response), we will not be able to use 'Expires: 0' for
this purpose.

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and Services
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216


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From daemon@optimus.ietf.org  Tue Jul 23 13:21:52 2002
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Date: Tue, 23 Jul 2002 09:44:58 -0700 (PDT)
From: Sean Olson <seancolson@yahoo.com>
Subject: Re: [Sip] Re: [Simple] -05 MESSAGE and Expires header
To: "Vijay K. Gurbani" <vkg@lucent.com>
Cc: hisham.khartabil@nokia.com, bcampbell@dynamicsoft.com,
        adam@dynamicsoft.com, sip@ietf.org
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I'm not necessarily disagreeing with you.
I think the problem is that people have different
notions of what "stale" means for a MESSAGE request.
Does stale simply mean the message is old or does
it imply that the message is invalid in some sense?
Would you render a "stale" MESSAGE to the user or
should the User Agent drop it on the floor?

I don't see how "Expires: 0" could be used to
retract an instant message. My interpretation
of "Expires: 0" is that if it hit an intermediary
which could not immediately deliver the MESSAGE
to the appropriate destination, it would not store
the MESSAGE and return a 202, rather it would 
return something like a 480 response.

/sean

--- "Vijay K. Gurbani" <vkg@lucent.com> wrote:
> Sean Olson wrote:
> > I fail to see why Expires: 0 means the message
> > will never expire. How does 0 equate to infinity?
> > 
> > An Expires: 0 from the sender, in my opinion,
> > should indicate that this message should be
> delivered
> > immediately or not at all. I don't think it should
> > be used for the sender (or gateway or proxy) to
> > indicate that a message is "stale".
> 
> Staleness is what 'Expires: 0' currently implies for
> REGISTER
> and SUBSCRIBE.  And I realize that I am fighting a
> loosing battle,
> but for all it is worth, it will be good to accord
> only that semantic
> to 'Expires: 0', and no more.
> 
> Now, before everyone flames me, I do realize that
> the MESSAGE I-D does
> not allow for this behavior, nor are there any
> requirements in
> section 5.4 of RFC 2779 to that effect.  But there
> have been many
> times when I have sent an email which I wish I could
> retract.  If
> in the future we want to do the same for stored IMs
> (i.e. ones that
> elicited a 202 response), we will not be able to use
> 'Expires: 0' for
> this purpose.
> 
> - vijay
> -- 
> Vijay K. Gurbani 
> vkg@{lucent.com,research.bell-labs.com,acm.org}
> Wireless Networks Group/Internet Software and
> Services
> Lucent Technologies/Bell Labs Innovations, 2000
> Lucent Lane, Rm 6G-440
> Naperville, Illinois 60566     Voice: +1 630 224
> 0216
> 


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From daemon@ns.ietf.org  Tue Jul 23 15:30:21 2002
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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: "'hisham.khartabil@nokia.com'" <hisham.khartabil@nokia.com>,
        seancolson@yahoo.com, bcampbell@dynamicsoft.com, vkg@lucent.com
Cc: adam@dynamicsoft.com, sip@ietf.org
Subject: RE: [Sip] Re: [Simple] -05 MESSAGE and Expires header
Date: Tue, 23 Jul 2002 19:53:15 +0100
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To the best of my knowledge, emergency calls have no more requirement for immediate delivery over and above any other SIP call.

There is a requirement of guarantee of reliable delivery to an appropriate emergency service answer point. In the case of local network congestion, all callers, including callers of the emergency service, will have equal difficulty in getting their INVITE through.

IEPREP is not dealing with emergency service, IEPREP is dealing with the ability of preferential authenticated callers to make calls at times of an emergency with preferential allocation of resources to make that call possible. These could potentially additionally be emergency service calls, but are more likely to be the chief of fire service calling local water supply chief or any other similar form of communication.

IEPREP is also meant to state requirements. Therefore any decision on what the protocol does in any particular condition, and what protocol is required to meet the requirements, rests with SIP, not with IEPREP, even when the issue is an IEPREP requirements statement.

Keith

> -----Original Message-----
> From: hisham.khartabil@nokia.com [mailto:hisham.khartabil@nokia.com]
> Sent: 23 July 2002 08:51
> To: seancolson@yahoo.com; bcampbell@dynamicsoft.com; vkg@lucent.com
> Cc: adam@dynamicsoft.com; sip@ietf.org
> Subject: RE: [Sip] Re: [Simple] -05 MESSAGE and Expires header
> 
> 
> I would like some interpretation for a few messages in terms 
> of immediate delivery, urgency and expiration of a message:
> 
> 1.
> MESSAGE ...
> Priority: non-urgent
> Expires: 0
> ...
> I'm leaving for lunch right now
> 
> 2.
> MESSAGE ...
> Priority: emergency
> Expires: 3600
> ...
> your house is on fire
> 
> 3.
> MESSAGE ...
> ...
> Date: Tue, 23 Jul 2002 7:25:00 GMT
> Expires: 60
> 
> 4.
> MESSAGE ...
> ...
> Date: Tue, 23 Jul 2002 7:25:00 GMT
> Expires: 0
> 
> 
> Do messages 1 and 2 get immediate delivery?
> 
> Another question: This immediate delivery debate is only 
> talking about Store and Forward type of servers of gateways. 
> What about other intermediaries? They certainly can't use the 
> expires-header to decide on speed of delivery of a message. 
> We need a new mechanism here. Again I ask the question: Is 
> this done in IEPREP?
> 
> Regards,
> Hisham
> 
> > -----Original Message-----
> > From: ext Sean Olson [mailto:seancolson@yahoo.com]
> > Sent: Tuesday, July 23, 2002 9:39 AM
> > To: Khartabil Hisham (NMP/Helsinki); bcampbell@dynamicsoft.com;
> > vkg@lucent.com
> > Cc: adam@dynamicsoft.com; sip@ietf.org
> > Subject: RE: [Sip] Re: [Simple] -05 MESSAGE and Expires header
> > 
> > 
> > I fail to see why Expires: 0 means the message
> > will never expire. How does 0 equate to infinity?
> > 
> > An Expires: 0 from the sender, in my opinion,
> > should indicate that this message should be delivered
> > immediately or not at all. I don't think it should
> > be used for the sender (or gateway or proxy) to
> > indicate that a message is "stale".
> > 
> > By stale,
> > I assume you mean a message that was intended for
> > immediate delivery, but was actually stored instead
> > -- this is not something existing SIP headers do a
> > good job of representing. I wouldn't try to hack
> > Expires: to fit this. Furthermore, I don't think
> > this is a problem that the message draft should have
> > to solve. Until someone comes up with a more complete
> > draft for offline message retrieval, I think we can
> > punt this. Or is that what you were offering to do?
> > 
> > 
> > --- hisham.khartabil@nokia.com wrote:
> > > So, a MESSAGE with Expires: 10 means it has 10
> > > seconds before it's stale while a MESSAGE with
> > > Expires: 0 means it will never be stale?
> > > 
> > > How do u inform the receiver that a message is
> > > stale?
> > > 
> > > /Hisham
> > > 
> > > > -----Original Message-----
> > > > From: ext Sean Olson [mailto:seancolson@yahoo.com]
> > > > Sent: Monday, July 22, 2002 8:07 PM
> > > > To: Ben Campbell; Vijay K. Gurbani
> > > > Cc: Adam Roach; SIP LIST
> > > > Subject: Re: [Sip] Re: [Simple] -05 MESSAGE and
> > > Expires header
> > > > 
> > > > 
> > > > So we have three things we might want to 
> > > > represent:
> > > > 
> > > > 1) Priority -- we use the Priority header for this
> > > > 2) Timeliness of content -- we use Expires: header
> > > >    for this
> > > > 3) Urgency -- we don't currently have a way to 
> > > >    represent this
> > > > 
> > > > Obviously 2 & 3 are related but not identical.
> > > > The message doesn't have to be urgent in order
> > > > for you to request immediate delivery. I think
> > > > solving (2) is enough for now. It is the more
> > > > interesting case for gateways and other
> > > > intermediaries that have to decide between a 
> > > > store-and-forward or immediate delivery approach.
> > > > In other words, stick with what we have today in
> > > > the draft.
> > > > 
> > > > /sean
> > > > 
> > > > __________________________________________________
> > > > Do You Yahoo!?
> > > > Yahoo! Health - Feel better, live better
> > > > http://health.yahoo.com
> > > > 
> > > > _______________________________________________
> > > > Sip mailing list 
> > > https://www1.ietf.org/mailman/listinfo/sip
> > > > This list is for NEW development of the core SIP
> > > Protocol
> > > > Use sip-implementors@cs.columbia.edu for questions
> > > on current sip
> > > > Use sipping@ietf.org for new developments on the
> > > application of sip
> > > > 
> > 
> > 
> > __________________________________________________
> > Do You Yahoo!?
> > Yahoo! Health - Feel better, live better
> > http://health.yahoo.com
> > 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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From daemon@ns.ietf.org  Tue Jul 23 16:12:02 2002
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From: "Vijay K. Gurbani" <vkg@lucent.com>
Organization: Lucent Technologies, Inc./Bell Laboratories
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To: Sean Olson <seancolson@yahoo.com>
CC: bcampbell@dynamicsoft.com, sip@ietf.org
Subject: Re: [Sip] Re: [Simple] -05 MESSAGE and Expires header
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Sean Olson wrote:
[...]
  > I don't see how "Expires: 0" could be used to
  > retract an instant message.

It can't right now; and there aren't any requirements for it
either.  However, by using 'Expires: 0' to mean immediate delivery
now, we may be closing the door on using it for retraction in the
future.  I am of the opinion that there is another header which can
provide immediate delivery (maybe not a best fit, but all the same, an
alternative to overloading 'Expires: 0').

Thanks,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and Services
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216











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From daemon@ns.ietf.org  Tue Jul 23 18:27:28 2002
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Date: Tue, 23 Jul 2002 17:07:31 -0500
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CC: hisham.khartabil@nokia.com, adam@dynamicsoft.com, dsardana@seven.com,
        vkg@lucent.com, Ya-Ching.Tan@icn.siemens.de,
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I concur. I have had many responses suggesting we leave it alone, and 
only a couple of people arguing for a change.

The draft currently leaves most of the handling of Expires as a matter 
of local policy. I think that is the appropriate level of detail for the 
MESSAGE specification. Since the majority of the controversy concerns 
store and forward relays, I will echo Sean's statement that if we choose 
to standardize such relays, that specification if free to further 
constrain relay behavior.

Also, as mentioned in the wg meeting, this discussion belongs on the SIP 
list, as it concerns a SIP work item. I have tried moving it there, but 
it doesn't seem to stick. Please direct any future discussion to the SIP 
list.

Jonathan Rosenberg wrote:
> I still believe that we should follow Robert's advice as offered in the 
> SIMPLE session. We are now down to arguing the interpretation of the 
> terms priority, urgency, and expiration. This is an argument for which 
> there is no right answer. Since 3261 allows each method to define a 
> method-specific meaning for Expires, we are well within the scope of the 
> specifications. No one has found a technical flaw in this approach. At 
> least several people (myself included) feel its reasonable if not 
> perfect. Robert's advice was to leave it be unless people found it to be 
> broken. I would rather finish this specification finally, rather than 
> debating the relative interpretations of the terms in play here.
> 
> -Jonathan R.
> 
> Ben Campbell wrote:
> 
>> hisham.khartabil@nokia.com wrote:
>>
>>> Well, you read my mind. I wanted to get consensus that Priority header
>>
>>
>> is for user interface purposes only before I do so.
>>
>> As I mentioned in a separate mail, priority _can_ be used for routing 
>> purposes. But it means something different than urgency.
>>
>>
>>> Should we do something like that or are these issues part of the
>>
>>
>> ieprep work?
>>
>>> Regards,
>>> Hisham
>>>
>>>
>>>
>>>> -----Original Message-----
>>>> From: ext Adam Roach [mailto:adam@dynamicsoft.com]
>>>> Sent: Monday, July 22, 2002 4:51 PM
>>>> To: Khartabil Hisham (NMP/Helsinki); Jonathan Rosenberg; Adam Roach
>>>> Cc: Ben Campbell; dsardana@seven.com; vkg@lucent.com;
>>>> Ya-Ching.Tan@icn.siemens.de; simple@mailman.dynamicsoft.com
>>>> Subject: RE: [Simple] -05 MESSAGE and Expires header
>>>>
>>>>
>>>>
>>>>
>>>>> -----Original Message-----
>>>>> From: hisham.khartabil@nokia.com [mailto:hisham.khartabil@nokia.com]
>>>>> - When is a message urgent but does not require immediate delivery?
>>>>
>>>>
>>>> Are you proposing an "Urgency" header, or confusing priority with
>>>> urgency? I'll see if I can accurately demonstrate their orthangonality.
>>>>
>>>> High Priority, Low Urgency:
>>>> Note to new residents: failure to file income taxes by April
>>>> 15th can lead to substantial penalties.
>>>>
>>>> Low Priority, High Urgency:
>>>> You have 5 minutes to come by and pick up the rest of your lunch
>>>> or I'm going to throw it away.
>>>>                            High Priority, High Urgency:
>>>> There's a tornado coming your way!
>>>>
>>>> Low Priority, Low Urgency:
>>>> Llamas can make four distinct noises: humming, clucking, orgling, and
>>>> a high pitched, rhythmic alarm sound.
>>>>
>>>> /a
>>>>
>>>
>>>
>>
>>
>>
> 
> 




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From daemon@ns.ietf.org  Tue Jul 23 18:34:29 2002
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I concur. I have had many responses suggesting we leave it alone, and
only a couple of people arguing for a change.

The draft currently leaves most of the handling of Expires as a matter
of local policy. I think that is the appropriate level of detail for the
MESSAGE specification. Since the majority of the controversy concerns
store and forward relays, I will echo Sean's statement that if we choose
to standardize such relays, that specification if free to further
constrain relay behavior.

Also, as mentioned in the wg meeting, this discussion belongs on the SIP
list, as it concerns a SIP work item. I have tried moving it there, but
it doesn't seem to stick. Please direct any future discussion to the SIP
list.

Jonathan Rosenberg wrote:
 > I still believe that we should follow Robert's advice as offered in the
 > SIMPLE session. We are now down to arguing the interpretation of the
 > terms priority, urgency, and expiration. This is an argument for which
 > there is no right answer. Since 3261 allows each method to define a
 > method-specific meaning for Expires, we are well within the scope of the
 > specifications. No one has found a technical flaw in this approach. At
 > least several people (myself included) feel its reasonable if not
 > perfect. Robert's advice was to leave it be unless people found it to be
 > broken. I would rather finish this specification finally, rather than
 > debating the relative interpretations of the terms in play here.
 >
 > -Jonathan R.
 >
 > Ben Campbell wrote:
 >
 >> hisham.khartabil@nokia.com wrote:
 >>
 >>> Well, you read my mind. I wanted to get consensus that Priority header
 >>
 >>
 >> is for user interface purposes only before I do so.
 >>
 >> As I mentioned in a separate mail, priority _can_ be used for routing
 >> purposes. But it means something different than urgency.
 >>
 >>
 >>> Should we do something like that or are these issues part of the
 >>
 >>
 >> ieprep work?
 >>
 >>> Regards,
 >>> Hisham
 >>>
 >>>
 >>>
 >>>> -----Original Message-----
 >>>> From: ext Adam Roach [mailto:adam@dynamicsoft.com]
 >>>> Sent: Monday, July 22, 2002 4:51 PM
 >>>> To: Khartabil Hisham (NMP/Helsinki); Jonathan Rosenberg; Adam Roach
 >>>> Cc: Ben Campbell; dsardana@seven.com; vkg@lucent.com;
 >>>> Ya-Ching.Tan@icn.siemens.de; simple@mailman.dynamicsoft.com
 >>>> Subject: RE: [Simple] -05 MESSAGE and Expires header
 >>>>
 >>>>
 >>>>
 >>>>
 >>>>> -----Original Message-----
 >>>>> From: hisham.khartabil@nokia.com [mailto:hisham.khartabil@nokia.com]
 >>>>> - When is a message urgent but does not require immediate delivery?
 >>>>
 >>>>
 >>>> Are you proposing an "Urgency" header, or confusing priority with
 >>>> urgency? I'll see if I can accurately demonstrate their 
orthangonality.
 >>>>
 >>>> High Priority, Low Urgency:
 >>>> Note to new residents: failure to file income taxes by April
 >>>> 15th can lead to substantial penalties.
 >>>>
 >>>> Low Priority, High Urgency:
 >>>> You have 5 minutes to come by and pick up the rest of your lunch
 >>>> or I'm going to throw it away.
 >>>>                            High Priority, High Urgency:
 >>>> There's a tornado coming your way!
 >>>>
 >>>> Low Priority, Low Urgency:
 >>>> Llamas can make four distinct noises: humming, clucking, orgling, and
 >>>> a high pitched, rhythmic alarm sound.
 >>>>
 >>>> /a
 >>>>
 >>>
 >>>
 >>
 >>
 >>
 >
 >





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From daemon@optimus.ietf.org  Wed Jul 24 08:06:25 2002
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Date: Wed, 24 Jul 2002 01:52:17 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Subject: [Sip] Changes from bis-09 to RFC 3261
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Folks,

There were approximately 150 changes made to bis-09 for the generation of
RFC 3261. Coming up with this list was an exercise in extremely
detailed and painful review of the document, something which took us a
great deal of time. The vast majority of these were typos, comma insertions,
indentation fixes, and so on. However, there were a few dozen changes
that reflect "bug fixes" which were reported between bis-09 and RFC
3261. Here is a summary of those of any significance whatsoever (you 
will find most to be trivial):

* The definition of header field was not quite right. We corrected it
to:

Header Field: A header field is a component of the SIP message header.
A header field can appear as one or more header field rows. Header
field rows consist of a header field name and zero or more header
field values. Multiple header field values on a given header field row
are separated by commas. Some header fields can only have a single
header field value, and as a result, always appear as a single header
field row.

* The definition and mechanisms for URL encoding were based on a
normative reference to RFC 1738. This RFC has been obsoleted by RFC
2396. The references were updated to point to RFC 2396, and RFC 1738
was removed from the list of references.

* The specification did not define the default language for text body
parts. HTTP specifies that it's ISO-8859-1. Our reference to HTTP 1.1
for the definition of the Content-Type header would lead you to
believe that this is the default for SIP too. However, since SIP uses
UTF-8, this default makes no sense. So, we have explicitly stated that
the default charset is UTF-8. You will find this text in section
7.4.1:

"SIP messages MAY contain binary bodies or body parts. When no explicit 
charset
parameter is provided by the sender, media subtypes of the "text"
type are defined to have a default charset value of "UTF-8". "

* There are constraints on when a UAS can be stateless. Stateless UAS
are really meant for handling unauthenticated requests, and are not a
general behavior. As an example, you can't really be a stateless
registrar. This is now explicitly called out in Section 8.2.7, first
paragraph:


"..the response and resends it, just as if it were replying to the first
instance of the request. A UAS cannot be stateless unless the request
processing for that method would always result in the same response if
the requests are identical. This rules out stateless registrars, for
example."


* It was pointed out that the rules for matching a CANCEL to its
transaction were not quite right if you went through the formal
process. We've corrected that, with the new text in Section 9.2,
paragraph 1:

"The TU determines the transaction to be
cancelled by taking the CANCEL request, and then assuming that the
request method is anything but CANCEL or ACK and applying the transaction"
                                       ^^^^^^


* Bis-09 defined the behavior of CANCEL on any method besides CANCEL
as a "no-op". However, it didn't define what no-op meant. So, we
removed the term no-op. The text, in Section 9.2, paragraph 4, now
reads:

"A CANCEL request has no impact on the processing of transactions with
any other method defined in this specification."


* The specification was inconsistent in its interpretation of a
missing Expires header and expires contact param in a REGISTER
request. One section said it meant the client means for 3600, the
other said it was a server defined minimum. With the new rules on the
inability of a server to reduce the client requested value, there
would be no way to have the client allow the server to choose unless
we went with the latter definition. So, we did. Section 10.2.1.1, last
paragraph, now reads:

"If neither mechanism for expressing a suggested expiration time is
present in a REGISTER, the client is indicating its desire for the
server to choose."


* In REGISTER requests, the "scope" of the q-value was defined to be
relative to "other bindings present in the REGISTER message or
existing within the location service of the registrar". Those are two
different scopes. Which is it? Its the location service within the
registrar. The text in Section 10.2.1.2 now reads:

"The "q" parameter indicates a relative
  preference for the particular Contact header field value compared to
  other bindings for this address-of-record."

* bis-09 was inconsistent in its failure handling for REGISTER between
Contact: * and a URI in the Contact header. In the former case, an out
of sequence REGISTER did not update the specific binding with the
ordering failure, but the others could be updated, and in the
latter case, the update failed. Since we have specified that all
request processing is atomic, the Contact: * treatment was not
right. Therefore, the text in bullet 6 of Section 10.3 was updated to
indicate that the update is aborted, and the request fails, in the
event of a sequence error.

* The wording in section 10.3, bullet 7, was inconsistent. On one
hand, it said the server had to use values when explicitly specified,
but also said that it could shorten the interval. This has been
clarified. The values in the Expires header or expires contact param
are the client requested expirations, and the server can shorten
those.

* A question arose as to whether a forked INVITE request resulted in
multiple offer/answer exchanges (one for each dialog established) or
just one offer/answer exchange. Its multiple. This has been clarified
in Section 13.2.1, paragraph 8:

"..parameters that apply to those means, and addresses for receiving
media from the offerer. An offer/answer exchange is within the context of a
dialog, so that if a SIP INVITE results in multiple dialogs, each is a
separate offer/answer exchange."

* Text in bis-09 seemed to imply that you could always generate a new
offer in a request as long as you had no outstanding offers. However,
thats true in addition to any method-specific rules, such as the ones
outline for UPDATE. This has been clarified in Section 13.2.1, bullet 4:

"After having sent or received an answer to the first offer, the UAC
MAY generate subsequent offers in requests based on rules specified
for that method, but only if it has received answers to any previous
offers, and has not sent any offers to which it hasn't gotten an
answer."

* Similarly, there was some text in bis-09 which led you to believe
that were only two possible offer/answer exchanges (INVITE/200 or
200/ACK) that were EVER possible. Thats not right. Its only the case
for elements compliant to bis alone. If you support UPDATE, there are
more possible. This is clarified in Section 13.2.1, 4th from last
paragraph:

"Concretely, the above rules specify two exchanges for UAs compliant
to this specification alone - the offer is in...."

* The rules for timer C in bis-09 might lead you to believe that
retransmits of the INVITE can refresh timer C. Thats not so; its only
provisional responses. This is now clarified in Section 13.3.1.1, next
to last paragraph:

"A proxy has the option of canceling
a transaction when there is a gap of 3 minutes between responses in a
transaction."


* There was some confusion reported at sipit over whether a branch ID
needs to be distinct for each transaction through a spiral. Many
implementations apparently were inserting the same branch ID in each
pass, which is not right. This has been clarified in Section 16.6,
bullet point 8, first paragraph:

"This implies that the proxy will compute its own branch
parameter, which will be globally unique for that branch, and
contain the requisite magic cookie. Note that this implies that
the branch parameter will be different for different instances
of a spiraled or looped request through a proxy."

* The recommended algorithm for computing a branch ID in Section 16.6,
bullet point 8, omitted the Via header in the hash. Its now been
included.
                               ^^^^^^^^^^

* The spec was inconsistent in UA and proxy handling of transport
failures. A proxy was supposed to return a 400, whilst a UA, a
503. Its been made consistent to 503. This results in a change in the
text of Section 16.9, which now reads:

"If the transport layer notifies a proxy of an error when it tries to
forward a request (see Section 18.4), the proxy MUST behave as if the
forwarded request received a 503 (Service Unavailable) response."


* It was reported that the processing of timer D in the transaction
layer was wrong. This is the timer that fires in the INVITE client
transaction after it has ceased waiting for retransmits of the final
response, in order to send an ACK. Bis-09 said to inform the TU of a
timeout. However, this is not a failure condition, so that makes no
sense. As such, Section 17.1.1.2, now reads:

"If timer D fires while the client transaction is in the "Completed"
state, the client transaction MUST move to the terminated state."
                                                                    ^
* Bis-09 defined a response retransmission as any response that
matches a transaction matched by a previous response. Thats true ONLY
for final responses. It doesn't work for provisional. Since there is
no normative text associated with the term "response retransmission",
we removed this definition entirely from Section 17.1.3.


* We clarified that, when moving to terminated state for client invite
transactions, you need to delete the transaction immediately.

* The rules in Section 17.2.3, for matching a request to a
transaction when the branch ID had the cookie, were not quite
right. In particular, they were wrong for
CANCEL. The rules were updated:

    The
    request matches a transaction if:

     1.  the branch parameter in the request
    is equal to the one in the top Via header field of the request that
    created the transaction, and

     2. the sent-by value in the top Via of the
    request is equal to the one in the request that created the
    transaction, and

     3. the method of the request matches the one that created
    the transaction, except for ACK, where the method of the request that
    created the transaction is INVITE.


* Section 18.1.1 led you to believe that if a request was too large
for UDP, and you resent it, it had to be with TCP. ACtually, it can be
with any congestion controlled transport (as defined by RFC 2914). We
also clarified that if you change transports, the Via transport value
needs to change.


* Section 19.1.4, 4th bullet, had made reference to characters in the
unsafe character set, indicating that those needed to be
escape-encoded in URLs. However, RFC 2396 doesn't define "unsafe"
characters - that is legacy from RFC 1738, which is obsoleted by
2396. So, the mention of unsafe characters was removed.

* It was pointed out that the lexical reordering rules for converting
a tel URL to a SIP URL could result in a user part that was not a
valid telephone-subscriber as defined in RFC 2806. The specific
problem is that 2806 requires the isdn-subaddress and post-dial
parameters to appear before any extension parameters. If you simply
order the parameters without excepting these two, you might place an
extension param before post-dial or isdn-subaddress, which is
illegal. This has been fixed, including an example which had this
problem. See Section 19.1.6.

* It was pointed out that the definition of "optional" headers in
table 2/3 only discussed UA insertion of the header. However, some
optional headers are only inserted by proxies (such as
Record-Route). So, the definition of optional in Section 20 was
updated to include any element, not just a UA.

* There were a bunch of places were the spec said "the syntax of this
header is defined in Section XXX of [HXXX], referring to RFC
2616. However, we incorporated most of that syntax into bis, so these
statements were no longer correct. They were removed.

* Rules for equality matching of From and To header fields were
absent, even though there were references elsewhere in the text to the
existence of such rules. They were added. See Sections 20.20 and
20.39.

* Section 20.22 indicated that the recommended value for Max-Forwards
is 70. Of course, that is the recommended INITIAL value. That was
clarified.

* The qop="auth" attribute was missing from the example
Proxy-Authenticate message in Section 20.27. It was added. It was also
missing from Section 20.44, and was added there too.

* The spec was inconsistent on the maximum value of the expires
Contact parameter and the Expires header field. They were made
consistent. The maximum value is (2**32)-1.

* Table 2/3 indicated that Retry-After was allowed in the 500 and 503
response codes, but Section 20.33 only mentioned the 503 response
code. Section 20.33 was updated to include the 500 code as well.

* Section 20.35, on the Server header, had an example which led you to
believe that the field was used by proxy servers. Its not - its just
UAS. The example was updated to fix that.

* Section 20.37, on the Unsupported header field, failed to mention
that a compact form was defined (the other headers with compact forms
explicitly mention it in their section 20's). A mention was added.

* The transaction matching algorithm requires rules for equality of
Via header fields, but none was defined. A rule was added, see Section
20.42.

* Section 21.4.15 indicated that inclusion of the unsupported
extensions in a 420 response was a SHOULD. This is inconsistent with
other text that said it was a MUST. Its a MUST. Section 21.4.15 was
updated to reflect that.

* Section 22.3 mentioned that "Proxy-Authentication" header, which
doesn't exist. Its "Proxy-Authenticate". It was fixed.

* Some text in Section 23.2, paragraph 3, was missing in bis-09
because of a latex conversion error. Specifically, the text should
have said that the body is structured as an S/MIME multipart/signed
CMS SignedData body. Its been fixed.

* Section 23.3 had an example with the Content-Transfer-Encoding
header in it. This header is no longer used, and so was removed.

* Section 24.1, last paragraph, message F2, was missing the tag in the
To field. It was added.

* Section 26.2.1 had a mention of the TLS_RSA_WITH_AES_128_CBC_SHA
ciphersuite, but had no reference to where such a suite is
defined. Its RFC 3268, and a reference to that was added.

* The BNF had this construct where there was a list of choices, and
one of the elements in the choice was optional:


       foo  =  bar / baz / [boo]

turns out, this is actually not right, since it allows for an empty
value for foo, which is not what is intended. Therefore, the text that
described this convention, and all usages of it in the BNF, were
removed.

* The BNF for the user part of the SIP/SIPS URI allowed for an empty
username and an @ sign; that is "sip:@foo.com" was valid according to
the BNF. Its not. The BNF for the SIP-URI and SIPS-URI constructions
was updated to correct this. Here is the change that was made (^
characters indicate where the change is):


OLD
    SIP-URI          =  "sip:" [ userinfo "@" ] hostport
                        uri-parameters [ headers ]
    SIPS-URI         =  "sips:" [ userinfo "@" ] hostport
                        uri-parameters [ headers ]
    userinfo         =  [ user / telephone-subscriber [ ":" password ]]
    user             =  *( unreserved / escaped / user-unreserved )

NEW
    SIP-URI          =  "sip:" [ userinfo ] hostport
                                         ^^  delete "@"
                        uri-parameters [ headers ]
    SIPS-URI         =  "sips:" [ userinfo ] hostport
                                          ^^ delete "@"
                        uri-parameters [ headers ]
    userinfo         =  ( user / telephone-subscriber ) [ ":" password ] "@"
                        ^                             ^                 ^^^^
    user             =  1*( unreserved / escaped / user-unreserved )
                        ^

* The Accept, Accept-Encoding, and Accept-Language and Allow BNF were all
updated to allow for the header to have an empty value.


* The BNF for rfc1123-date was wrong. It was missing a space after the
comma. It was corrected thusly:

OLD
rfc1123-date  =  wkday "," date1 SP time SP "GMT"

NEW
rfc1123-date  =  wkday "," SP date1 SP time SP "GMT"
                            ^^

* The BNF for the Timestamp header was wrong. It was missing an
explicit LWS between the the send time and the delay. It was
added. The BNF was changed thusly:


OLD
Timestamp  =  "Timestamp" HCOLON 1*(DIGIT)
               [ "." *(DIGIT) ] [ delay ]

NEW
Timestamp  =  "Timestamp" HCOLON 1*(DIGIT)
               [ "." *(DIGIT) ] [ LWS delay ]
                                  ^^^


THanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


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From daemon@optimus.ietf.org  Wed Jul 24 08:32:58 2002
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Thread-Topic: [Simple] -05 MESSAGE and Expires header
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Subject: [Sip] RE: [Simple] -05 MESSAGE and Expires header
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We don't want the expires-header to be used as a source for routing decisions. We also don't want to special case "expires: 0".

Why should routing depend on the value on the expires header?

I quote Jonathan
"The general notion is that it indicates a lifetime of 
something; a point after which the thing is no longer valid."

You to have some general meaning of expires, otherwise I can say expires of 0 for a new method FOO means "don't FOO me back" or "explode your phone". If the general notion is what Jonathan said above (and what Adam said below), then Expires: 0 in a MESSAGE means that the MESSAGE expiration occurs in zero seconds. It doesn't mean immediate delivery.

I think the best thing to do is to remove the text. Other methods don't define the meaning of "Expires: 0".

Regards,
Hisham

> -----Original Message-----
> From: ext Adam Roach [mailto:adam@dynamicsoft.com]
> Sent: Wednesday, July 24, 2002 4:09 AM
> To: 'Bobby Sardana'; Adam Roach
> Cc: Jonathan Rosenberg; Ben Campbell; Khartabil Hisham (NMP/Helsinki);
> vkg@lucent.com; Ya-Ching.Tan@icn.siemens.de;
> simple@mailman.dynamicsoft.com
> Subject: RE: [Simple] -05 MESSAGE and Expires header
> 
> 
> > -----Original Message-----
> > From: Bobby Sardana [mailto:dsardana@seven.com]
> >
> > The proposal is simple: (not sure from a complete 
> > applicability point of
> > view, but here it is)
> > 
> > a. Expire signifies a unit of time
> 
> Agreed.
> 
> > b. For methods that have the Expire header, the unit of time 
> > determines
> > the method validitity.
> 
> There's a problem with that approach. For SUBSCRIBE and REGISTER,
> the "Expires" header doesn't indicate the period of validity of
> the request (which is presumably what you mean by "method validity");
> it indicates the period of validity for what is *created* by the
> request. The analog for INVITE would be "Session-Expires".
> 
> > The only exception here is 0, which can mean
> > "expired" or "infinite".
> 
> There are actually three problems here. 
> 
> 1. Historically, "0" has never been treated 
>    as a special case in SIP expirations. (More
>    on this below).
> 
> 2. "Expires: 0" does not mean "expired" (that
>    would, in theory, be represented by a negative
>    number -- but we don't do that); it means that
>    expiration occurs in zero seconds (i.e.
>    instantaneously). For the purposes of instant
>    messages, that's exactly what we want to convey:
>    "this mesage expires instantaneously -- so, send
>    it on or consider it stale" (where handling of
>    stale messages is a matter of local policy).
> 
> 3. Zero has *never* meant infinity in a SIP
>    "Expires" context. The convention for "infinity"
>    is to use 4294967295 (2^32-1), which ends up being
>    over 136 years -- long enough to mean "infinity" for
>    the purposes of a computer protocol.
> 
> /a
> 

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From daemon@optimus.ietf.org  Wed Jul 24 08:35:32 2002
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To: Sachin Shenoy <sachins@netlab.hcltech.com>, sip@ietf.org
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Subject: [Sip] Re: [Sip-implementors] Loop detection & Via's branch
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Indeed. Using previous branch in calculating next branch
disables loop detection. Sounds like a spec inconsistency.

-Jiri

(moved to SIP mailing list)

At 09:47 AM 7/24/2002, Sachin Shenoy wrote:
>Hi,
>
>In RFC 3261 [16.6 (8)] it says that, for loop detection at proxy, 
>we should calculate the branch using top most Via header also. 
>I guess while using top-most Via header we should avoid using the branch. If branch is also used, the loop can go undetected as 
>the branch could be unique, even when the request has looped.
>(8.1.1.7 Says branch has to be unique across space and time).
>
>Hope I got it right. If I have missed anything please let me
>know.
>
>Thanks
>Sachin
>
>
>16.6 (8)
>"This is to ensure that if the
>request is routed back to the proxy and one of those fields
>changes, it is treated as a spiral and not a loop (see Section
>16.3).  A common way to create this value is to compute a
>cryptographic hash of the To tag, From tag, Call-ID header
>field, the Request-URI of the request received (before
>translation), *the topmost Via header*, and the sequence number
>from the CSeq header field, in addition to any Proxy-Require
>and Proxy-Authorization header fields that may be present." 


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Subject: [Sip] I-D ACTION:draft-ietf-sip-refer-06.txt
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--NextPart

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

	Title		: The SIP Refer Method
	Author(s)	: R. Sparks
	Filename	: draft-ietf-sip-refer-06.txt
	Pages		: 21
	Date		: 23-Jul-02
	
This document defines the REFER method.  This SIP extension requests
that the recipient REFER to a resource provided in the request.  It
provides a mechanism allowing the party sending the REFER to be
notified of the outcome of the referenced request.  This can be used
to enable many applications, including call transfer.
In addition to the REFER method, this document defines the the refer
event package and the Refer-To request header.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-refer-06.txt

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From daemon@optimus.ietf.org  Wed Jul 24 10:03:02 2002
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Thread-Topic: [Sip] Re: [Simple] -05 MESSAGE and Expires header
Thread-Index: AcIybaQ8Nqh0SFiMT4CBtyDZS/gnkAAqQTIQ
From: "ShavinderPal Singh" <ShavinderPal.Singh@ssdsi.com>
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Hi,

Can anyone help me in finding the Grammar for the Refer-To and
Referred-By headers, which are present in the REFER message?
I could not find their mention in RFC 3261.

Thanks & Regards
s

Ph.: 091-124-6383474 (e 250)
__________________________________________
I had a good grip on life, but then the handle broke.



-----Original Message-----
From: Sean Olson [mailto:seancolson@yahoo.com]
Sent: Tuesday, July 23, 2002 10:15 PM
To: Vijay K. Gurbani
Cc: hisham.khartabil@nokia.com; bcampbell@dynamicsoft.com;
adam@dynamicsoft.com; sip@ietf.org
Subject: Re: [Sip] Re: [Simple] -05 MESSAGE and Expires header


I'm not necessarily disagreeing with you.
I think the problem is that people have different
notions of what "stale" means for a MESSAGE request.
Does stale simply mean the message is old or does
it imply that the message is invalid in some sense?
Would you render a "stale" MESSAGE to the user or
should the User Agent drop it on the floor?

I don't see how "Expires: 0" could be used to
retract an instant message. My interpretation
of "Expires: 0" is that if it hit an intermediary
which could not immediately deliver the MESSAGE
to the appropriate destination, it would not store
the MESSAGE and return a 202, rather it would 
return something like a 480 response.

/sean

--- "Vijay K. Gurbani" <vkg@lucent.com> wrote:
> Sean Olson wrote:
> > I fail to see why Expires: 0 means the message
> > will never expire. How does 0 equate to infinity?
> > 
> > An Expires: 0 from the sender, in my opinion,
> > should indicate that this message should be
> delivered
> > immediately or not at all. I don't think it should
> > be used for the sender (or gateway or proxy) to
> > indicate that a message is "stale".
> 
> Staleness is what 'Expires: 0' currently implies for
> REGISTER
> and SUBSCRIBE.  And I realize that I am fighting a
> loosing battle,
> but for all it is worth, it will be good to accord
> only that semantic
> to 'Expires: 0', and no more.
> 
> Now, before everyone flames me, I do realize that
> the MESSAGE I-D does
> not allow for this behavior, nor are there any
> requirements in
> section 5.4 of RFC 2779 to that effect.  But there
> have been many
> times when I have sent an email which I wish I could
> retract.  If
> in the future we want to do the same for stored IMs
> (i.e. ones that
> elicited a 202 response), we will not be able to use
> 'Expires: 0' for
> this purpose.
> 
> - vijay
> -- 
> Vijay K. Gurbani 
> vkg@{lucent.com,research.bell-labs.com,acm.org}
> Wireless Networks Group/Internet Software and
> Services
> Lucent Technologies/Bell Labs Innovations, 2000
> Lucent Lane, Rm 6G-440
> Naperville, Illinois 60566     Voice: +1 630 224
> 0216
> 


__________________________________________________
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http://health.yahoo.com

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Wed Jul 24 10:23:02 2002
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'ShavinderPal Singh'" <ShavinderPal.Singh@ssdsi.com>, sip@ietf.org
Subject: RE: [Sip] Grammar for Refer-To and Referred-By
Date: Wed, 24 Jul 2002 09:53:35 -0400
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http://www.ietf.org/internet-drafts/draft-ietf-sip-refer-06.txt

> -----Original Message-----
> From: ShavinderPal Singh [mailto:ShavinderPal.Singh@ssdsi.com]
> Sent: Wednesday, July 24, 2002 9:35 AM
> To: sip@ietf.org
> Subject: [Sip] Grammar for Refer-To and Referred-By
> 
> 
> Hi,
> 
> Can anyone help me in finding the Grammar for the Refer-To and
> Referred-By headers, which are present in the REFER message?
> I could not find their mention in RFC 3261.
> 
> Thanks & Regards
> s
> 
> Ph.: 091-124-6383474 (e 250)
> __________________________________________
> I had a good grip on life, but then the handle broke.
> 
> 
> 
> -----Original Message-----
> From: Sean Olson [mailto:seancolson@yahoo.com]
> Sent: Tuesday, July 23, 2002 10:15 PM
> To: Vijay K. Gurbani
> Cc: hisham.khartabil@nokia.com; bcampbell@dynamicsoft.com;
> adam@dynamicsoft.com; sip@ietf.org
> Subject: Re: [Sip] Re: [Simple] -05 MESSAGE and Expires header
> 
> 
> I'm not necessarily disagreeing with you.
> I think the problem is that people have different
> notions of what "stale" means for a MESSAGE request.
> Does stale simply mean the message is old or does
> it imply that the message is invalid in some sense?
> Would you render a "stale" MESSAGE to the user or
> should the User Agent drop it on the floor?
> 
> I don't see how "Expires: 0" could be used to
> retract an instant message. My interpretation
> of "Expires: 0" is that if it hit an intermediary
> which could not immediately deliver the MESSAGE
> to the appropriate destination, it would not store
> the MESSAGE and return a 202, rather it would 
> return something like a 480 response.
> 
> /sean
> 
> --- "Vijay K. Gurbani" <vkg@lucent.com> wrote:
> > Sean Olson wrote:
> > > I fail to see why Expires: 0 means the message
> > > will never expire. How does 0 equate to infinity?
> > > 
> > > An Expires: 0 from the sender, in my opinion,
> > > should indicate that this message should be
> > delivered
> > > immediately or not at all. I don't think it should
> > > be used for the sender (or gateway or proxy) to
> > > indicate that a message is "stale".
> > 
> > Staleness is what 'Expires: 0' currently implies for
> > REGISTER
> > and SUBSCRIBE.  And I realize that I am fighting a
> > loosing battle,
> > but for all it is worth, it will be good to accord
> > only that semantic
> > to 'Expires: 0', and no more.
> > 
> > Now, before everyone flames me, I do realize that
> > the MESSAGE I-D does
> > not allow for this behavior, nor are there any
> > requirements in
> > section 5.4 of RFC 2779 to that effect.  But there
> > have been many
> > times when I have sent an email which I wish I could
> > retract.  If
> > in the future we want to do the same for stored IMs
> > (i.e. ones that
> > elicited a 202 response), we will not be able to use
> > 'Expires: 0' for
> > this purpose.
> > 
> > - vijay
> > -- 
> > Vijay K. Gurbani 
> > vkg@{lucent.com,research.bell-labs.com,acm.org}
> > Wireless Networks Group/Internet Software and
> > Services
> > Lucent Technologies/Bell Labs Innovations, 2000
> > Lucent Lane, Rm 6G-440
> > Naperville, Illinois 60566     Voice: +1 630 224
> > 0216
> > 
> 
> 
> __________________________________________________
> Do You Yahoo!?
> Yahoo! Health - Feel better, live better
> http://health.yahoo.com
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From daemon@optimus.ietf.org  Wed Jul 24 10:54:30 2002
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Reply-To: "Ranjit Avasarala" <ranjit.avasarala@bplmail.com>
From: "Ranjit Avasarala" <ranjit.avasarala@bplmail.com>
To: "ShavinderPal Singh" <ShavinderPal.Singh@ssdsi.com>, <sip@ietf.org>
References: <89CFEE6120A0224EA07781892EC0B48B9040B9@esuims01.esamsungumit.com>
Subject: Re: [Sip] Grammar for Refer-To and Referred-By
Date: Wed, 24 Jul 2002 19:59:10 +0530
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For Refer-To header refer  draft-ietf-sip-refer-06.txt
and for referred-by refer draft-ietf-sip-referredby-00.txt

HTH
Ranjit
----- Original Message ----- 
From: ShavinderPal Singh <ShavinderPal.Singh@ssdsi.com>
To: <sip@ietf.org>
Sent: Wednesday, July 24, 2002 7:05 PM
Subject: [Sip] Grammar for Refer-To and Referred-By


> Hi,
> 
> Can anyone help me in finding the Grammar for the Refer-To and
> Referred-By headers, which are present in the REFER message?
> I could not find their mention in RFC 3261.
> 
> Thanks & Regards
> s
> 
> Ph.: 091-124-6383474 (e 250)
> __________________________________________
> I had a good grip on life, but then the handle broke.
> 
> 
> 
> -----Original Message-----
> From: Sean Olson [mailto:seancolson@yahoo.com]
> Sent: Tuesday, July 23, 2002 10:15 PM
> To: Vijay K. Gurbani
> Cc: hisham.khartabil@nokia.com; bcampbell@dynamicsoft.com;
> adam@dynamicsoft.com; sip@ietf.org
> Subject: Re: [Sip] Re: [Simple] -05 MESSAGE and Expires header
> 
> 
> I'm not necessarily disagreeing with you.
> I think the problem is that people have different
> notions of what "stale" means for a MESSAGE request.
> Does stale simply mean the message is old or does
> it imply that the message is invalid in some sense?
> Would you render a "stale" MESSAGE to the user or
> should the User Agent drop it on the floor?
> 
> I don't see how "Expires: 0" could be used to
> retract an instant message. My interpretation
> of "Expires: 0" is that if it hit an intermediary
> which could not immediately deliver the MESSAGE
> to the appropriate destination, it would not store
> the MESSAGE and return a 202, rather it would 
> return something like a 480 response.
> 
> /sean
> 
> --- "Vijay K. Gurbani" <vkg@lucent.com> wrote:
> > Sean Olson wrote:
> > > I fail to see why Expires: 0 means the message
> > > will never expire. How does 0 equate to infinity?
> > > 
> > > An Expires: 0 from the sender, in my opinion,
> > > should indicate that this message should be
> > delivered
> > > immediately or not at all. I don't think it should
> > > be used for the sender (or gateway or proxy) to
> > > indicate that a message is "stale".
> > 
> > Staleness is what 'Expires: 0' currently implies for
> > REGISTER
> > and SUBSCRIBE.  And I realize that I am fighting a
> > loosing battle,
> > but for all it is worth, it will be good to accord
> > only that semantic
> > to 'Expires: 0', and no more.
> > 
> > Now, before everyone flames me, I do realize that
> > the MESSAGE I-D does
> > not allow for this behavior, nor are there any
> > requirements in
> > section 5.4 of RFC 2779 to that effect.  But there
> > have been many
> > times when I have sent an email which I wish I could
> > retract.  If
> > in the future we want to do the same for stored IMs
> > (i.e. ones that
> > elicited a 202 response), we will not be able to use
> > 'Expires: 0' for
> > this purpose.
> > 
> > - vijay
> > -- 
> > Vijay K. Gurbani 
> > vkg@{lucent.com,research.bell-labs.com,acm.org}
> > Wireless Networks Group/Internet Software and
> > Services
> > Lucent Technologies/Bell Labs Innovations, 2000
> > Lucent Lane, Rm 6G-440
> > Naperville, Illinois 60566     Voice: +1 630 224
> > 0216
> > 
> 
> 
> __________________________________________________
> Do You Yahoo!?
> Yahoo! Health - Feel better, live better
> http://health.yahoo.com
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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From: Adam Roach <adam@dynamicsoft.com>
To: "'hisham.khartabil@nokia.com'" <hisham.khartabil@nokia.com>,
        Adam Roach <adam@dynamicsoft.com>, sip@ietf.org
Subject: RE: [Sip] RE: [Simple] -05 MESSAGE and Expires header
Date: Wed, 24 Jul 2002 11:03:06 -0500
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> -----Original Message-----
> From: hisham.khartabil@nokia.com [mailto:hisham.khartabil@nokia.com]
> 
> We don't want the expires-header to be used as a source for 
> routing decisions. We also don't want to special case "expires: 0".
> 
> Why should routing depend on the value on the expires header?

Proxies can route on whatever makes sense for them. If I want
a proxy to route based on whether some string appears in the
Subject field, I can. If I want it to route based on a hash
of the SDP, I can. If I want it to route based on the second
digit of the hour portion of the date field, I can. You can
argue that such things might not be useful (and I'll agree)
but a proxy can make routing decisions based on whatever
makes sense for whatever service they are designed to provide.

So, if I (as a service provider) want to deploy a node that
bases its behavior on the "Expires" header -- either because
it makes sense for me, or provides a wanted service for my
users -- then that's what I'm going to do. I don't think
the MESSAGE spec needs to mention this in any way, though.

(As an analogy, 3261 does not mention anything about making
routing decisions based on the "Accept-Language" field --
but you can certainly imagine circumstances in which it would
make sense to do so).

> I think the best thing to do is to remove the text. Other 
> methods don't define the meaning of "Expires: 0".

Except section 10.2.2 of RFC 3261 (REGISTER).

And section 3.1.4.3 of RFC 3265 (SUBSCRIBE).

While "Expires: 0" isn't a special case, it is (obviously)
cause for confusion (otherwise we wouldn't have a huge
discussion thread spanning two mailing lists arguing about
its meaning). So, from a purely theoretical perspective,
your point has some validity: the meaning of "Expires: 0"
*can* be deduced using rigorous and consistent thinking.

However, from a *practical* perspective, it's obvious that
some people would get the semantics of "Expires: 0" wrong
unless there is some guidance provided -- and varying
semantics will harm interoperability. I argue for a
consistent handling, but explicit text the clarifies
the issue to leave no room for misinterpretation.

I further argue that the current document already satisifies
this request.

/a

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From daemon@optimus.ietf.org  Wed Jul 24 12:53:28 2002
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Jonathan Rosenberg wrote:
[...]
  > Robert's advice was to leave it be unless people found it to be
  > broken. I would rather finish this specification finally, rather than
  > debating the relative interpretations of the terms in play here.

Jonathan/Ben:

It was not my intention to delay the specification when I originally
posted this question to the SIMPLE list in Yokohama (heck, I did not
even think that it would engender such passion :-)).  Out of a set of
9 people who have contributed to this thread, 1/3 (including me) think
that 'Expires: 0' should NOT be used.  So that settles the issue in
favor of keeping status quo.

  > No one has found a technical flaw in this approach

These are the technical flaws as I see them, none of which are show
stoppers:

        (1) Expires is generally considered to be an e2e header; however,
            MESSAGE I-D uses it for decision making in intermediaries (I
            will note that even in RFC 3261, it is suggested that proxies
            not use Expires header in determining when to teardown a
            session that is being set up)
        (2) Expires will need to be duplicated in the 'outer' headers if
            S/MIME is being used

The philosophical flaws are probably more egregious; these are:

        (1) Violates the self-describing aspect of SIP headers by
            overloading 'Expires: 0' to imply immediate delivery
        (2) Non-congruence of usage with respect to REGISTER and
            SUBSCRIBE
        (3) Leaves open the possibility that future requests will
            interpret 'Expires: 0' in other unknown subtle ways
        (4) Why overload it if there is another header that
            approximates immediate delivery to both intermediaries
            and end points
        (5) While IMs send in session mode probably cannot be with-
            drawn; maybe in the future IMs in page mode could be.
            For example, an IM with an 'Expires: 60' which elicits a
            202 response from a relay, could be retracted.  If so, we
            loose the ability to use 'Expires: 0' for retraction
Thanks,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Wireless Networks Group/Internet Software and Services
Lucent Technologies/Bell Labs Innovations, 2000 Lucent Lane, Rm 6G-440
Naperville, Illinois 60566     Voice: +1 630 224 0216


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From daemon@optimus.ietf.org  Wed Jul 24 13:57:14 2002
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CC: Henning Schulzrinne <hgs@cs.columbia.edu>, SIP-WG <sip@ietf.org>
Subject: Re: [Sip] language info in SIP headers
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Shingo Fujimoto wrote:
> I am happy to hear you already had some discussions.

Henning and I were trying to determine the "right" thing to
to regarding language tagging, and found, as you are
finding, that it is a far from simple answer....

specific comment inline:


> 
> Henning Schulzrinne wrote:
> 
> 
>>- RFC 2482 or http://www.unicode.org/unicode/reports/tr27/#tag provide
>>language tagging, as you noted. While I have not seen an official
>>statement from the IESG on this, I believe 10646 language tagging has
>>generally been deprecated. It's not quite clear whether the reasons (I
>>believe, nesting) for that apply to short text such as Subject or
>>Organization.
> 
> 
> 
> You are probably right.
> 
> Here is the comment from Ned Freed, one of the RFC2231 authors:
> 
> 
>>Language indicators in a pure UTF-8 environment like SIP headers have
> 
> to be
> 
>>stored as part of the UTF-8 string itself. The plane 14 language tags
> 
> described
> 
>>in
>>
>>   http://www.unicode.org/unicode/reports/tr27/#tag
>>
>>are the only means of doing this that I'm aware of. However, the use
>>of such tags is, as the text indicates, strongly discouraged.
>>
>>It is the Unicode Technical Committee's belief that
>>
>>  The requirement for language information embedded in plain text data
> 
> is often
> 
>>  overstated.
>>
>>As such, I see little chance for support for such tagging in the
> 
> context of SIP
> 
>>headers being widely deployed.
>>
>>				Ned


I think the point Ned is making here, is that language
tagging is not all that useful in practice. With UTF-8,
you can render Kanji characters. You need the language
tagging IFF some automata needs to do something special
based on knowledge of the language, such as translation
or text to speech. However, it is far from clear whether
that information is truly needed. Translation of names
(like those in Subject, From, To) is not likely to work
in any case, nor is it clear that the language tag is
needed for it...



> The reason I prefer this approach is that SIP messages must be small
> if it runs over UDP. "language tag" will add only few bytes.
> 

Well, as Ned indicated, and Henning repeated, the in-band
language tagging provided by UTF-8 is not considered
acceptable, and we would not be able to write a specification
which recommends it as the solution for sip.

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


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From daemon@optimus.ietf.org  Wed Jul 24 15:17:26 2002
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CC: Shingo Fujimoto <shingo_fujimoto@jp.fujitsu.com>, SIP-WG <sip@ietf.org>,
        Ned.Freed@innosoft.com
Subject: Re: [Sip] language info in SIP headers
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> I think the point Ned is making here, is that language
> tagging is not all that useful in practice. With UTF-8,
> you can render Kanji characters. You need the language
> tagging IFF some automata needs to do something special
> based on knowledge of the language, such as translation
> or text to speech. However, it is far from clear whether
> that information is truly needed. Translation of names
> (like those in Subject, From, To) is not likely to work
> in any case, nor is it clear that the language tag is
> needed for it...

You're about to step into the Han unification minefield. From my limited
understanding, the same Unicode/10646 code point is rendered differently
depending on the language (Japanese, Chinese, etc.). I don't know if
this is also true for text-to-speech. Certainly, text-to-speech is
useful for all three headers, e.g., in a voicemail system. Jonathan
Rosenberg ;lang=de and Jonathan Rosenberg ;lang=en sound very different.

> Well, as Ned indicated, and Henning repeated, the in-band
> language tagging provided by UTF-8 is not considered
> acceptable, and we would not be able to write a specification
> which recommends it as the solution for sip.

I think he meant the ?=jp? marking, not UTF-8 language tagging. (That
said, it isn't clear to me why this would be inherently better, except
maybe to avoid tripping non-language-tag-capable 10646 renderers and
annoy the user with random characters instead... The arguments against
tagging seem to be about the same in either syntax - nesting, and all that.)

A pragmatic choice is to say that To/From don't need UTF or ?=? tagging,
since ;language=jp will work just fine, without annoying users with
random ASCII art. The Subject header is more difficult, but there, a
smart text-to-speech renderer can usually guess at the language by
checking its dictionary. From my vague recollection, very few words are
needed to do fairly accurate language recognition, excepting deliberate
attempts to create phrases that are meaningful in two languages. Google
seems to do an ok guessing job on most web pages, for example. (Guessing
isn't pretty, but many users aren't going to bother marking their
Subject line anyway and routinely use multiple languages, so that device
configuration isn't sufficient.)

> 
> -Jonathan R.


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From daemon@optimus.ietf.org  Wed Jul 24 17:18:44 2002
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From: Sean Olson <seancolson@yahoo.com>
Subject: Re: [Sip] language info in SIP headers
To: Henning Schulzrinne <hgs@cs.columbia.edu>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: Shingo Fujimoto <shingo_fujimoto@jp.fujitsu.com>, SIP-WG <sip@ietf.org>,
        Ned.Freed@innosoft.com
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Is there a problem to be solved here that cannot
be handled by Accept-Language and Content-Language?
I realize these only cover the payload and not the
headers. My assumption is that most of the information
that you would need to extract from the headers
can either be treated as a string of octets by an
automata 
-or-
is for human consumption and therefore it is
sufficient if these can be rendered appropriately
to the end user (ignoring the Han unification 
problem for now) which UTF-8 should cover. Proper
names as in the From/To are a good example of this.

What automata application cannot make use of the 
Accept-Language header to do its job? (I'm not
saying one does not exist, I would just like
clarification of the problem we are trying to solve 
here)

Regards,
Sean Olson

--- Henning Schulzrinne <hgs@cs.columbia.edu> wrote:
> > I think the point Ned is making here, is that
> language
> > tagging is not all that useful in practice. With
> UTF-8,
> > you can render Kanji characters. You need the
> language
> > tagging IFF some automata needs to do something
> special
> > based on knowledge of the language, such as
> translation
> > or text to speech. However, it is far from clear
> whether
> > that information is truly needed. Translation of
> names
> > (like those in Subject, From, To) is not likely to
> work
> > in any case, nor is it clear that the language tag
> is
> > needed for it...
> 
> You're about to step into the Han unification
> minefield. From my limited
> understanding, the same Unicode/10646 code point is
> rendered differently
> depending on the language (Japanese, Chinese, etc.).
> I don't know if
> this is also true for text-to-speech. Certainly,
> text-to-speech is
> useful for all three headers, e.g., in a voicemail
> system. Jonathan
> Rosenberg ;lang=de and Jonathan Rosenberg ;lang=en
> sound very different.
> 
> > Well, as Ned indicated, and Henning repeated, the
> in-band
> > language tagging provided by UTF-8 is not
> considered
> > acceptable, and we would not be able to write a
> specification
> > which recommends it as the solution for sip.
> 
> I think he meant the ?=jp? marking, not UTF-8
> language tagging. (That
> said, it isn't clear to me why this would be
> inherently better, except
> maybe to avoid tripping non-language-tag-capable
> 10646 renderers and
> annoy the user with random characters instead... The
> arguments against
> tagging seem to be about the same in either syntax -
> nesting, and all that.)
> 
> A pragmatic choice is to say that To/From don't need
> UTF or ?=? tagging,
> since ;language=jp will work just fine, without
> annoying users with
> random ASCII art. The Subject header is more
> difficult, but there, a
> smart text-to-speech renderer can usually guess at
> the language by
> checking its dictionary. From my vague recollection,
> very few words are
> needed to do fairly accurate language recognition,
> excepting deliberate
> attempts to create phrases that are meaningful in
> two languages. Google
> seems to do an ok guessing job on most web pages,
> for example. (Guessing
> isn't pretty, but many users aren't going to bother
> marking their
> Subject line anyway and routinely use multiple
> languages, so that device
> configuration isn't sufficient.)
> 
> > 
> > -Jonathan R.
> 
> 
> _______________________________________________
> Sip mailing list 
> https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP
> Protocol
> Use sip-implementors@cs.columbia.edu for questions
> on current sip
> Use sipping@ietf.org for new developments on the
> application of sip


__________________________________________________
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_______________________________________________
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From daemon@optimus.ietf.org  Wed Jul 24 17:41:57 2002
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Date: Wed, 24 Jul 2002 17:13:30 -0400
From: Henning Schulzrinne <hgs@cs.columbia.edu>
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To: Sean Olson <seancolson@yahoo.com>
CC: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        Shingo Fujimoto <shingo_fujimoto@jp.fujitsu.com>,
        SIP-WG <sip@ietf.org>, Ned.Freed@innosoft.com
Subject: Re: [Sip] language info in SIP headers
References: <20020724204608.49648.qmail@web11602.mail.yahoo.com>
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I don't think this is a major issue, but there is a semantic difference. 
Accept-Language lists *all* the languages I'm willing to deal with, in 
terms of error messages, so it might be something like

Accept-Language: en, de, fr

This doesn't tell the receiver text-to-speech engine anything about how 
to pronounce my name. As noted, for personal names, even the same ASCII 
string (ignoring the much more complicated CJK sets) can be pronounced 
quite differently, particularly where the name has been "imported". I 
believe the problem is more pronounced with CJK since Chinese names are 
rendered in "classical" Chinese, rather than simplified. (And I'm 
probably getting the details wrong.)


Sean Olson wrote:
> Is there a problem to be solved here that cannot
> be handled by Accept-Language and Content-Language?
> I realize these only cover the payload and not the
> headers. My assumption is that most of the information
> that you would need to extract from the headers
> can either be treated as a string of octets by an
> automata 
> -or-
> is for human consumption and therefore it is
> sufficient if these can be rendered appropriately
> to the end user (ignoring the Han unification 
> problem for now) which UTF-8 should cover. Proper
> names as in the From/To are a good example of this.
> 
> What automata application cannot make use of the 
> Accept-Language header to do its job? (I'm not
> saying one does not exist, I would just like
> clarification of the problem we are trying to solve 
> here)
> 
> Regards,
> Sean Olson
> 
> --- Henning Schulzrinne <hgs@cs.columbia.edu> wrote:
> 
>>>I think the point Ned is making here, is that
>>
>>language
>>
>>>tagging is not all that useful in practice. With
>>
>>UTF-8,
>>
>>>you can render Kanji characters. You need the
>>
>>language
>>
>>>tagging IFF some automata needs to do something
>>
>>special
>>
>>>based on knowledge of the language, such as
>>
>>translation
>>
>>>or text to speech. However, it is far from clear
>>
>>whether
>>
>>>that information is truly needed. Translation of
>>
>>names
>>
>>>(like those in Subject, From, To) is not likely to
>>
>>work
>>
>>>in any case, nor is it clear that the language tag
>>
>>is
>>
>>>needed for it...
>>
>>You're about to step into the Han unification
>>minefield. From my limited
>>understanding, the same Unicode/10646 code point is
>>rendered differently
>>depending on the language (Japanese, Chinese, etc.).
>>I don't know if
>>this is also true for text-to-speech. Certainly,
>>text-to-speech is
>>useful for all three headers, e.g., in a voicemail
>>system. Jonathan
>>Rosenberg ;lang=de and Jonathan Rosenberg ;lang=en
>>sound very different.
>>
>>
>>>Well, as Ned indicated, and Henning repeated, the
>>
>>in-band
>>
>>>language tagging provided by UTF-8 is not
>>
>>considered
>>
>>>acceptable, and we would not be able to write a
>>
>>specification
>>
>>>which recommends it as the solution for sip.
>>
>>I think he meant the ?=jp? marking, not UTF-8
>>language tagging. (That
>>said, it isn't clear to me why this would be
>>inherently better, except
>>maybe to avoid tripping non-language-tag-capable
>>10646 renderers and
>>annoy the user with random characters instead... The
>>arguments against
>>tagging seem to be about the same in either syntax -
>>nesting, and all that.)
>>
>>A pragmatic choice is to say that To/From don't need
>>UTF or ?=? tagging,
>>since ;language=jp will work just fine, without
>>annoying users with
>>random ASCII art. The Subject header is more
>>difficult, but there, a
>>smart text-to-speech renderer can usually guess at
>>the language by
>>checking its dictionary. From my vague recollection,
>>very few words are
>>needed to do fairly accurate language recognition,
>>excepting deliberate
>>attempts to create phrases that are meaningful in
>>two languages. Google
>>seems to do an ok guessing job on most web pages,
>>for example. (Guessing
>>isn't pretty, but many users aren't going to bother
>>marking their
>>Subject line anyway and routinely use multiple
>>languages, so that device
>>configuration isn't sufficient.)
>>
>>
>>>-Jonathan R.
>>
>>
>>_______________________________________________
>>Sip mailing list 
>>https://www1.ietf.org/mailman/listinfo/sip
>>This list is for NEW development of the core SIP
>>Protocol
>>Use sip-implementors@cs.columbia.edu for questions
>>on current sip
>>Use sipping@ietf.org for new developments on the
>>application of sip
> 
> 
> 
> __________________________________________________
> Do You Yahoo!?
> Yahoo! Health - Feel better, live better
> http://health.yahoo.com


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Wed Jul 24 18:38:11 2002
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From: "Shan Lu" <shanlu@sentito.com>
To: "IETF SIP Mailing List" <sip@ietf.org>
Date: Wed, 24 Jul 2002 18:14:43 -0400
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Hi,

RFC3261 says server must not challenge ACK. It also says that UACs
_will_ duplicate Authorization header of INVITE in ACK. I believe this
"will" strength is too weak. Think of a stateless proxy that performs
authentication. If it receives an ACK with no credentials, it knows it
should not challenge the ACK. But what does it do? It has two options
and it will have a hard time figuring out when to do what:

1. Drop it. But an ACK may be legitimately without credentials (like ACK
for 200) and should be sent along.

2. Proxy it. But the INVITE may have been challenged (ACK for 407) and
the UAS will receive ACK out of the blue. Not detrimental but certainly
not nice.

So I think the "will"  requirement on UAC to include same Authorization
header as INVITE in ACK needs to be strengthened to "MUST" level.

Regards,

Shan Lu

sentitO Networks


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From daemon@optimus.ietf.org  Wed Jul 24 19:17:27 2002
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Date: Wed, 24 Jul 2002 17:55:19 -0500
From: Ben Campbell <bcampbell@dynamicsoft.com>
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To: hisham.khartabil@nokia.com
CC: adam@dynamicsoft.com, sip@ietf.org
Subject: Compromise on Expires header in Message (was Re: [Sip] RE: [Simple]
 -05 MESSAGE and Expires header)
References: <2038BCC78B1AD641891A0D1AE133DBB7C20400@esebe019.NOE.Nokia.com>
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I suggest as a compromise, we simply strike any reference to a special 
case for Expires:0 from the draft. It really belongs in a specification 
for store-and-forward relays, and is probably not in scope for the 
MESSAGE specification.

If anyone cares to write a draft for store and forward relays, I will be 
happy to argue the point then.

The draft has already completed last call, but this seems to be a valid 
piece of last call feedback. In the interest of getting an update out 
prior to IESG review, I am submitting this change in an 06 version now.



hisham.khartabil@nokia.com wrote:
> We don't want the expires-header to be used as a source for routing decisions. We also don't want to special case "expires: 0".
> 
> Why should routing depend on the value on the expires header?
> 
> I quote Jonathan
> "The general notion is that it indicates a lifetime of 
> something; a point after which the thing is no longer valid."
> 
> You to have some general meaning of expires, otherwise I can say expires of 0 for a new method FOO means "don't FOO me back" or "explode your phone". If the general notion is what Jonathan said above (and what Adam said below), then Expires: 0 in a MESSAGE means that the MESSAGE expiration occurs in zero seconds. It doesn't mean immediate delivery.
> 
> I think the best thing to do is to remove the text. Other methods don't define the meaning of "Expires: 0".
> 
> Regards,
> Hisham
> 
> 
>>-----Original Message-----
>>From: ext Adam Roach [mailto:adam@dynamicsoft.com]
>>Sent: Wednesday, July 24, 2002 4:09 AM
>>To: 'Bobby Sardana'; Adam Roach
>>Cc: Jonathan Rosenberg; Ben Campbell; Khartabil Hisham (NMP/Helsinki);
>>vkg@lucent.com; Ya-Ching.Tan@icn.siemens.de;
>>simple@mailman.dynamicsoft.com
>>Subject: RE: [Simple] -05 MESSAGE and Expires header
>>
>>
>>
>>>-----Original Message-----
>>>From: Bobby Sardana [mailto:dsardana@seven.com]
>>>
>>>The proposal is simple: (not sure from a complete 
>>>applicability point of
>>>view, but here it is)
>>>
>>>a. Expire signifies a unit of time
>>
>>Agreed.
>>
>>
>>>b. For methods that have the Expire header, the unit of time 
>>>determines
>>>the method validitity.
>>
>>There's a problem with that approach. For SUBSCRIBE and REGISTER,
>>the "Expires" header doesn't indicate the period of validity of
>>the request (which is presumably what you mean by "method validity");
>>it indicates the period of validity for what is *created* by the
>>request. The analog for INVITE would be "Session-Expires".
>>
>>
>>>The only exception here is 0, which can mean
>>>"expired" or "infinite".
>>
>>There are actually three problems here. 
>>
>>1. Historically, "0" has never been treated 
>>   as a special case in SIP expirations. (More
>>   on this below).
>>
>>2. "Expires: 0" does not mean "expired" (that
>>   would, in theory, be represented by a negative
>>   number -- but we don't do that); it means that
>>   expiration occurs in zero seconds (i.e.
>>   instantaneously). For the purposes of instant
>>   messages, that's exactly what we want to convey:
>>   "this mesage expires instantaneously -- so, send
>>   it on or consider it stale" (where handling of
>>   stale messages is a matter of local policy).
>>
>>3. Zero has *never* meant infinity in a SIP
>>   "Expires" context. The convention for "infinity"
>>   is to use 4294967295 (2^32-1), which ends up being
>>   over 136 years -- long enough to mean "infinity" for
>>   the purposes of a computer protocol.
>>
>>/a
>>
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 




_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From daemon@optimus.ietf.org  Wed Jul 24 19:25:50 2002
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From: Adam Roach <adam@dynamicsoft.com>
To: Ben Campbell <bcampbell@dynamicsoft.com>, hisham.khartabil@nokia.com
Cc: Adam Roach <adam@dynamicsoft.com>, sip@ietf.org
Subject: RE: Compromise on Expires header in Message (was Re: [Sip] RE: [S
	imple] -05 MESSAGE and Expires header)
Date: Wed, 24 Jul 2002 17:58:30 -0500
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I agree with Ben's approach.

/a

> -----Original Message-----
> From: Ben Campbell [mailto:bcampbell@dynamicsoft.com]
> Sent: Wednesday, July 24, 2002 17:55
> To: hisham.khartabil@nokia.com
> Cc: adam@dynamicsoft.com; sip@ietf.org
> Subject: Compromise on Expires header in Message (was Re: [Sip] RE:
> [Simple] -05 MESSAGE and Expires header)
> 
> 
> I suggest as a compromise, we simply strike any reference to 
> a special 
> case for Expires:0 from the draft. It really belongs in a 
> specification 
> for store-and-forward relays, and is probably not in scope for the 
> MESSAGE specification.
> 
> If anyone cares to write a draft for store and forward 
> relays, I will be 
> happy to argue the point then.
> 
> The draft has already completed last call, but this seems to 
> be a valid 
> piece of last call feedback. In the interest of getting an update out 
> prior to IESG review, I am submitting this change in an 06 
> version now.
> 
> 
> 
> hisham.khartabil@nokia.com wrote:
> > We don't want the expires-header to be used as a source for 
> routing decisions. We also don't want to special case "expires: 0".
> > 
> > Why should routing depend on the value on the expires header?
> > 
> > I quote Jonathan
> > "The general notion is that it indicates a lifetime of 
> > something; a point after which the thing is no longer valid."
> > 
> > You to have some general meaning of expires, otherwise I 
> can say expires of 0 for a new method FOO means "don't FOO me 
> back" or "explode your phone". If the general notion is what 
> Jonathan said above (and what Adam said below), then Expires: 
> 0 in a MESSAGE means that the MESSAGE expiration occurs in 
> zero seconds. It doesn't mean immediate delivery.
> > 
> > I think the best thing to do is to remove the text. Other 
> methods don't define the meaning of "Expires: 0".
> > 
> > Regards,
> > Hisham
> > 
> > 
> >>-----Original Message-----
> >>From: ext Adam Roach [mailto:adam@dynamicsoft.com]
> >>Sent: Wednesday, July 24, 2002 4:09 AM
> >>To: 'Bobby Sardana'; Adam Roach
> >>Cc: Jonathan Rosenberg; Ben Campbell; Khartabil Hisham 
> (NMP/Helsinki);
> >>vkg@lucent.com; Ya-Ching.Tan@icn.siemens.de;
> >>simple@mailman.dynamicsoft.com
> >>Subject: RE: [Simple] -05 MESSAGE and Expires header
> >>
> >>
> >>
> >>>-----Original Message-----
> >>>From: Bobby Sardana [mailto:dsardana@seven.com]
> >>>
> >>>The proposal is simple: (not sure from a complete 
> >>>applicability point of
> >>>view, but here it is)
> >>>
> >>>a. Expire signifies a unit of time
> >>
> >>Agreed.
> >>
> >>
> >>>b. For methods that have the Expire header, the unit of time 
> >>>determines
> >>>the method validitity.
> >>
> >>There's a problem with that approach. For SUBSCRIBE and REGISTER,
> >>the "Expires" header doesn't indicate the period of validity of
> >>the request (which is presumably what you mean by "method 
> validity");
> >>it indicates the period of validity for what is *created* by the
> >>request. The analog for INVITE would be "Session-Expires".
> >>
> >>
> >>>The only exception here is 0, which can mean
> >>>"expired" or "infinite".
> >>
> >>There are actually three problems here. 
> >>
> >>1. Historically, "0" has never been treated 
> >>   as a special case in SIP expirations. (More
> >>   on this below).
> >>
> >>2. "Expires: 0" does not mean "expired" (that
> >>   would, in theory, be represented by a negative
> >>   number -- but we don't do that); it means that
> >>   expiration occurs in zero seconds (i.e.
> >>   instantaneously). For the purposes of instant
> >>   messages, that's exactly what we want to convey:
> >>   "this mesage expires instantaneously -- so, send
> >>   it on or consider it stale" (where handling of
> >>   stale messages is a matter of local policy).
> >>
> >>3. Zero has *never* meant infinity in a SIP
> >>   "Expires" context. The convention for "infinity"
> >>   is to use 4294967295 (2^32-1), which ends up being
> >>   over 136 years -- long enough to mean "infinity" for
> >>   the purposes of a computer protocol.
> >>
> >>/a
> >>
> > 
> > 
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> > 
> 
> 
> 

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From daemon@optimus.ietf.org  Wed Jul 24 19:32:18 2002
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To: Ben Campbell <bcampbell@dynamicsoft.com>
CC: hisham.khartabil@nokia.com, adam@dynamicsoft.com, sip@ietf.org
Subject: Re: Compromise on Expires header in Message (was Re: [Sip] RE: [Simple]
 -05 MESSAGE and Expires header)
References: <2038BCC78B1AD641891A0D1AE133DBB7C20400@esebe019.NOE.Nokia.com> <3D3F3057.4060202@dynamicsoft.com>
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(Replies to self...)

Well, after sending this note, I read more carefully and realized that 
this course of action is exactly what Hisham was suggesting. Therefore 
please amend the first sentence to read "I suggest we accept Hisham's 
proposal as a compromise..."

Ben Campbell wrote:
> I suggest as a compromise, we simply strike any reference to a special 
> case for Expires:0 from the draft. It really belongs in a specification 
> for store-and-forward relays, and is probably not in scope for the 
> MESSAGE specification.
> 
> If anyone cares to write a draft for store and forward relays, I will be 
> happy to argue the point then.
> 
> The draft has already completed last call, but this seems to be a valid 
> piece of last call feedback. In the interest of getting an update out 
> prior to IESG review, I am submitting this change in an 06 version now.
> 
> 
> 
> hisham.khartabil@nokia.com wrote:
> 
>> We don't want the expires-header to be used as a source for routing 
>> decisions. We also don't want to special case "expires: 0".
>>
>> Why should routing depend on the value on the expires header?
>>
>> I quote Jonathan
>> "The general notion is that it indicates a lifetime of something; a 
>> point after which the thing is no longer valid."
>>
>> You to have some general meaning of expires, otherwise I can say 
>> expires of 0 for a new method FOO means "don't FOO me back" or 
>> "explode your phone". If the general notion is what Jonathan said 
>> above (and what Adam said below), then Expires: 0 in a MESSAGE means 
>> that the MESSAGE expiration occurs in zero seconds. It doesn't mean 
>> immediate delivery.
>>
>> I think the best thing to do is to remove the text. Other methods 
>> don't define the meaning of "Expires: 0".
>>
>> Regards,
>> Hisham
>>
>>
>>> -----Original Message-----
>>> From: ext Adam Roach [mailto:adam@dynamicsoft.com]
>>> Sent: Wednesday, July 24, 2002 4:09 AM
>>> To: 'Bobby Sardana'; Adam Roach
>>> Cc: Jonathan Rosenberg; Ben Campbell; Khartabil Hisham (NMP/Helsinki);
>>> vkg@lucent.com; Ya-Ching.Tan@icn.siemens.de;
>>> simple@mailman.dynamicsoft.com
>>> Subject: RE: [Simple] -05 MESSAGE and Expires header
>>>
>>>
>>>
>>>> -----Original Message-----
>>>> From: Bobby Sardana [mailto:dsardana@seven.com]
>>>>
>>>> The proposal is simple: (not sure from a complete applicability 
>>>> point of
>>>> view, but here it is)
>>>>
>>>> a. Expire signifies a unit of time
>>>
>>>
>>> Agreed.
>>>
>>>
>>>> b. For methods that have the Expire header, the unit of time determines
>>>> the method validitity.
>>>
>>>
>>> There's a problem with that approach. For SUBSCRIBE and REGISTER,
>>> the "Expires" header doesn't indicate the period of validity of
>>> the request (which is presumably what you mean by "method validity");
>>> it indicates the period of validity for what is *created* by the
>>> request. The analog for INVITE would be "Session-Expires".
>>>
>>>
>>>> The only exception here is 0, which can mean
>>>> "expired" or "infinite".
>>>
>>>
>>> There are actually three problems here.
>>> 1. Historically, "0" has never been treated   as a special case in 
>>> SIP expirations. (More
>>>   on this below).
>>>
>>> 2. "Expires: 0" does not mean "expired" (that
>>>   would, in theory, be represented by a negative
>>>   number -- but we don't do that); it means that
>>>   expiration occurs in zero seconds (i.e.
>>>   instantaneously). For the purposes of instant
>>>   messages, that's exactly what we want to convey:
>>>   "this mesage expires instantaneously -- so, send
>>>   it on or consider it stale" (where handling of
>>>   stale messages is a matter of local policy).
>>>
>>> 3. Zero has *never* meant infinity in a SIP
>>>   "Expires" context. The convention for "infinity"
>>>   is to use 4294967295 (2^32-1), which ends up being
>>>   over 136 years -- long enough to mean "infinity" for
>>>   the purposes of a computer protocol.
>>>
>>> /a
>>>
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>
> 
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 




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To: Henning Schulzrinne <hgs@cs.columbia.edu>
CC: SIP-WG <sip@ietf.org>
Subject: Re: [Sip] language info in SIP headers
References: <3D3CE1C9.4080505@jp.fujitsu.com> <3D3D0C1A.7020307@cs.columbia.edu> <3D3D22C6.2080208@jp.fujitsu.com> <3D3D268B.8030803@cs.columbia.edu>
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Henning Schulzrinne wrote:

> 
> I would suggest you write up a short I-D on the subject.
> 


Yes, I will.


I just wonder if I should write it as a requirement doc, or just a specification proposal doc.

Because I think the requirements are some kind of simple, e.g. UTF-8 
friendly and multiple language in same field.

-- 
======================================
          SHINGO FUJIMOTO
     FUJITSU LABORATORIES LIMITED
E-mail: shingo_fujimoto@jp.fujitsu.com



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From daemon@optimus.ietf.org  Thu Jul 25 01:10:33 2002
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Subject: [Sip] nit for rfc 3261
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I am not sure if the following has already been reported as a bug/nit
(wrong use of the word terminate):

RFC 3261:
6 Definitions

Final Response: A response that terminates a SIP transaction,


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From daemon@optimus.ietf.org  Thu Jul 25 02:05:54 2002
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To: "Shan Lu" <shanlu@sentito.com>, "IETF SIP Mailing List" <sip@ietf.org>
From: Jiri Kuthan <kuthan@fokus.gmd.de>
Subject: Re: [Sip] Authentication and ACK
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Actually, there is no security in ACK unless you use secure transport.
Resending INVITE credentials in ACK can be easily attacked by a replay
attack. Whichever requirement strength you'ld like to put on sending
credentials in ACK, don't expect security to improve.

-Jiri

At 12:14 AM 7/25/2002, Shan Lu wrote:
>Hi,
>
>RFC3261 says server must not challenge ACK. It also says that UACs
>_will_ duplicate Authorization header of INVITE in ACK. I believe this
>"will" strength is too weak. Think of a stateless proxy that performs
>authentication. If it receives an ACK with no credentials, it knows it
>should not challenge the ACK. But what does it do? It has two options
>and it will have a hard time figuring out when to do what:
>
>1. Drop it. But an ACK may be legitimately without credentials (like ACK
>for 200) and should be sent along.
>
>2. Proxy it. But the INVITE may have been challenged (ACK for 407) and
>the UAS will receive ACK out of the blue. Not detrimental but certainly
>not nice.
>
>So I think the "will"  requirement on UAC to include same Authorization
>header as INVITE in ACK needs to be strengthened to "MUST" level.
>
>Regards,
>
>Shan Lu
>
>sentitO Networks
>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip 


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Subject: [Sip] some comments on draft-ietf-sip-session-timer-09.txt
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Hi,

some questions and comments on draft-ietf-sip-session-timer-09.txt. Pardon
me if these have already been discussed before.

1.Why is it that only a session refresh request or its 2xx response
supposed to refresh the session?(See section 10. Performing Refreshes)
For example consider the following scenario:
UAC                                                             UAS
                session is established
                with UAS as the refresher
(session timer active)                          (session timer active)


(UAC wishes to change some parameters of the session)
(UAC sends an re-INVITE)
 --------INV---------------------------------------->
                                                                    (Should
UAS not reset its session timer on receiving it?)
<-------200 Ok-----------------------------------
(Should UAC also not reset its session timer on receiving 200 OK?)


2.Section 8.1 Processing of Requests says
"If the request contains a Supported header field with a value
   "timer", the proxy MAY reject the INVITE request with a 422 (Session
   Interval Too Small) response if the session interval in the Session-
   Expires header field is smaller than the minimum interval defined by
   the proxy's local policy. When sending the 422 response, the proxy
   MUST include a Min-SE header field with the value of its minimum
   interval.".

Apart from this option, does the draft allow the proxy to  increase its
local minimum interval?.
Can the proxy also forward the request?
Can it discard the request, send no response back?if the proxy does not
respond back with a 422,
the requestor/refresher will not know what value to use for the session
timer in its retry.
The draft does not specify what (other) actions can the proxy take in this
case.
If there are no other possible useful actions for the proxy, then should
not  the MAY be changed to SHOULD?

regards
harsh







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From daemon@optimus.ietf.org  Thu Jul 25 03:03:24 2002
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To: Jiri Kuthan <kuthan@fokus.gmd.de>
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Subject: Re: [Sip] Authentication and ACK
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Hi,
     If the Authorization header is duplicated from INVITE to ACK,
is it that servers are not supposed to treat such requests
as replay attacks even though a previous request arrived with
the exact same nonce-count ? This is fine
in case of stateful proxies, but are stateless proxies also
expected to maintain such state information to determine whether
the ACK credentials are valid ?

Rgds
Seshu




Jiri Kuthan <kuthan@fokus.gmd.de> on 07/25/2002 11:52:00 AM

To:   "Shan Lu" <shanlu@sentito.com>, "IETF SIP Mailing List"
      <sip@ietf.org>
cc:    (bcc: Seshashayi T/HSSBLR)

Subject:  Re: [Sip] Authentication and ACK




Actually, there is no security in ACK unless you use secure transport.
Resending INVITE credentials in ACK can be easily attacked by a replay
attack. Whichever requirement strength you'ld like to put on sending
credentials in ACK, don't expect security to improve.

-Jiri

At 12:14 AM 7/25/2002, Shan Lu wrote:
>Hi,
>
>RFC3261 says server must not challenge ACK. It also says that UACs
>_will_ duplicate Authorization header of INVITE in ACK. I believe this
>"will" strength is too weak. Think of a stateless proxy that performs
>authentication. If it receives an ACK with no credentials, it knows it
>should not challenge the ACK. But what does it do? It has two options
>and it will have a hard time figuring out when to do what:
>
>1. Drop it. But an ACK may be legitimately without credentials (like ACK
>for 200) and should be sent along.
>
>2. Proxy it. But the INVITE may have been challenged (ACK for 407) and
>the UAS will receive ACK out of the blue. Not detrimental but certainly
>not nice.
>
>So I think the "will"  requirement on UAC to include same Authorization
>header as INVITE in ACK needs to be strengthened to "MUST" level.
>
>Regards,
>
>Shan Lu
>
>sentitO Networks
>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip





_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From daemon@optimus.ietf.org  Thu Jul 25 04:29:58 2002
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Subject: RE: [Sip] Authentication and ACK
Date: Thu, 25 Jul 2002 08:39:16 +0100
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Thread-Topic: [Sip] Authentication and ACK
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From: "James Undery" <jundery@ubiquity.net>
To: <seshashayi@hss.hns.com>, "Jiri Kuthan" <kuthan@fokus.gmd.de>
Cc: "Shan Lu" <shanlu@sentito.com>, "IETF SIP Mailing List" <sip@ietf.org>
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You also have to remember that ACKs for non-2xx final responses are generated hop by hop and so can't contain any "useful" authentication. (Unless everyone in the network can authenticate you ;-)

James

> -----Original Message-----
> From: seshashayi@hss.hns.com [mailto:seshashayi@hss.hns.com]
> Sent: 25 July 2002 07:17
> To: Jiri Kuthan
> Cc: Shan Lu; IETF SIP Mailing List
> Subject: Re: [Sip] Authentication and ACK
> 
> 
> 
> 
> Hi,
>      If the Authorization header is duplicated from INVITE to ACK,
> is it that servers are not supposed to treat such requests
> as replay attacks even though a previous request arrived with
> the exact same nonce-count ? This is fine
> in case of stateful proxies, but are stateless proxies also
> expected to maintain such state information to determine whether
> the ACK credentials are valid ?
> 
> Rgds
> Seshu
> 
> 
> 
> 
> Jiri Kuthan <kuthan@fokus.gmd.de> on 07/25/2002 11:52:00 AM
> 
> To:   "Shan Lu" <shanlu@sentito.com>, "IETF SIP Mailing List"
>       <sip@ietf.org>
> cc:    (bcc: Seshashayi T/HSSBLR)
> 
> Subject:  Re: [Sip] Authentication and ACK
> 
> 
> 
> 
> Actually, there is no security in ACK unless you use secure transport.
> Resending INVITE credentials in ACK can be easily attacked by a replay
> attack. Whichever requirement strength you'ld like to put on sending
> credentials in ACK, don't expect security to improve.
> 
> -Jiri
> 
> At 12:14 AM 7/25/2002, Shan Lu wrote:
> >Hi,
> >
> >RFC3261 says server must not challenge ACK. It also says that UACs
> >_will_ duplicate Authorization header of INVITE in ACK. I 
> believe this
> >"will" strength is too weak. Think of a stateless proxy that performs
> >authentication. If it receives an ACK with no credentials, 
> it knows it
> >should not challenge the ACK. But what does it do? It has two options
> >and it will have a hard time figuring out when to do what:
> >
> >1. Drop it. But an ACK may be legitimately without 
> credentials (like ACK
> >for 200) and should be sent along.
> >
> >2. Proxy it. But the INVITE may have been challenged (ACK 
> for 407) and
> >the UAS will receive ACK out of the blue. Not detrimental 
> but certainly
> >not nice.
> >
> >So I think the "will"  requirement on UAC to include same 
> Authorization
> >header as INVITE in ACK needs to be strengthened to "MUST" level.
> >
> >Regards,
> >
> >Shan Lu
> >
> >sentitO Networks
> >
> >
> >_______________________________________________
> >Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >This list is for NEW development of the core SIP Protocol
> >Use sip-implementors@cs.columbia.edu for questions on current sip
> >Use sipping@ietf.org for new developments on the application of sip
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 
> 
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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From daemon@optimus.ietf.org  Thu Jul 25 08:41:16 2002
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Subject: [Sip] I-D ACTION:draft-willis-sip-congestsafe-00.txt
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: An Extension to the Session Initiation protocol to 
                          Assure Congestion Safety
	Author(s)	: D. Willis, B. Campbell
	Filename	: draft-willis-sip-congestsafe-00.txt
	Pages		: 11
	Date		: 24-Jul-02
	
The Session Initiation Protocol allows the use of UDP for transport
of SIP messages.  The use of UDP inherently risks network congestion
problems, as UDP itself does not define congestion prevention,
avoidance, detection, or correction mechanisms.  This problem is
aggravated by large SIP messages which fragment at the UDP level.
Transport protocols in SIP are also negotiated on a per-hop basis, at
the SIP level, so SIP proxies may convert from TCP to UDP and so
forth.  This document defines what it means for SIP nodes to be
congestion safe and specifies an extension by which a SIP User Agent
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From daemon@optimus.ietf.org  Thu Jul 25 09:09:06 2002
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From: BECKMANN MARK <mark.beckmann@sal.siemens.de>
To: "'sip@ietf.org'" <sip@ietf.org>
Date: Thu, 25 Jul 2002 14:43:25 +0200
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Subject: [Sip] Comments on draft-rosenberg-sip-reg-00
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Hi Jonathan,

in your draft on the event package for registrations, you proposed the
following FSM for the contacts registered with an adress of record:
                                                                        
                                                                          
                                 +------+                                 
                                 |      | refreshed                       
                                 |      |                                 
                                 V      |                                 
    +------------+            +------------+            +------------+    
    |            |            |            |            |            |    
    |    Init    |----------->|   Active   |----------->| Terminated |    
    |            |            |            |            |            |    
    +------------+ registered +------------+ expired    +------------+    
                   created                   deactivated                  
                                             probation                    
                                             unregistered                 
                                             rejected                     
                                                                          

In section 3.1 it is described that in case an administrator wants a device
to re-authenticate, the registration is terminated and the device is
notified, e.g. by setting the event parameter to "deactivated".

First of all, I can't see a big difference between the two events "expired"
and "deactivated". "Deactivated" seems to be part of the "expired" event. In
both cases the contact may be re-registered. It doesn't seem very important
whether the registration is moved to the terminated state because it has
expired or because it has been deactivated.

Secondly, I think in case an administrator wants a device to
re-authenticate, the registration of the corresponding contact should not
move to the terminated state. Instead, another event (for example
"re-authenticate") similar to the "refresh" event should be introduced. When
this event occurs, the device is notified and the duration for which the
contact remains registered is reduced and if the device fails to
re-register, the registration is moved to the terminated state e.g. via the
"expired" event.

Thus, the FSM would look like:
                                                                          
                                 +------+                                 
                                 |      | refreshed                       
                                 |      | re-authenticate

                                 V      |                                 
    +------------+            +------------+            +------------+    
    |            |            |            |            |            |    
    |    Init    |----------->|   Active   |----------->| Terminated |    
    |            |            |            |            |            |    
    +------------+ registered +------------+            +------------+    
                   created                   expired               
                                             probation            
                                             unregistered           
                                             rejected               

Best regards,


Mark


Mark Beckmann			Siemens AG

ICM MP P PS 4S2
P.O.Box 100702			phone: +49 (5341) 906 1814
D-38228 Salzgitter   		fax:   +49 (5341) 906 2010

mailto: Mark.Beckmann@siemens.com



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From: "Shan Lu" <shanlu@sentito.com>
To: "'James Undery'" <jundery@ubiquity.net>, <seshashayi@hss.hns.com>,
        "'Jiri Kuthan'" <kuthan@fokus.gmd.de>
Cc: "'IETF SIP Mailing List'" <sip@ietf.org>
Subject: RE: [Sip] Authentication and ACK
Date: Thu, 25 Jul 2002 09:39:34 -0400
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I don't think hop by hop is a problem. From the perspective of
authentication server/stateless proxy, the ACK is either for an INVITE
that has passed authentication or for an INVITE that has failed
authentication. Any stateful entities that ACK non-2xx final responses
would have no difficulty duplicating the same Authorization header, if
any, from INVITE. 

I agree it is subject to replay attack. But isn't it better than blindly
proxy each and every ACK that is received at stateless server ?

Regards,

Shan Lu

>-----Original Message-----
>From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf 
>Of James Undery
>Sent: Thursday, July 25, 2002 3:39 AM
>To: seshashayi@hss.hns.com; Jiri Kuthan
>Cc: Shan Lu; IETF SIP Mailing List
>Subject: RE: [Sip] Authentication and ACK
>
>
>You also have to remember that ACKs for non-2xx final 
>responses are generated hop by hop and so can't contain any 
>"useful" authentication. (Unless everyone in the network can 
>authenticate you ;-)
>
>James
>
>> -----Original Message-----
>> From: seshashayi@hss.hns.com [mailto:seshashayi@hss.hns.com]
>> Sent: 25 July 2002 07:17
>> To: Jiri Kuthan
>> Cc: Shan Lu; IETF SIP Mailing List
>> Subject: Re: [Sip] Authentication and ACK
>> 
>> 
>> 
>> 
>> Hi,
>>      If the Authorization header is duplicated from INVITE to ACK,
>> is it that servers are not supposed to treat such requests
>> as replay attacks even though a previous request arrived with
>> the exact same nonce-count ? This is fine
>> in case of stateful proxies, but are stateless proxies also
>> expected to maintain such state information to determine whether
>> the ACK credentials are valid ?
>> 
>> Rgds
>> Seshu
>> 
>> 
>> 
>> 
>> Jiri Kuthan <kuthan@fokus.gmd.de> on 07/25/2002 11:52:00 AM
>> 
>> To:   "Shan Lu" <shanlu@sentito.com>, "IETF SIP Mailing List"
>>       <sip@ietf.org>
>> cc:    (bcc: Seshashayi T/HSSBLR)
>> 
>> Subject:  Re: [Sip] Authentication and ACK
>> 
>> 
>> 
>> 
>> Actually, there is no security in ACK unless you use secure 
>transport.
>> Resending INVITE credentials in ACK can be easily attacked 
>by a replay
>> attack. Whichever requirement strength you'ld like to put on sending
>> credentials in ACK, don't expect security to improve.
>> 
>> -Jiri
>> 
>> At 12:14 AM 7/25/2002, Shan Lu wrote:
>> >Hi,
>> >
>> >RFC3261 says server must not challenge ACK. It also says that UACs
>> >_will_ duplicate Authorization header of INVITE in ACK. I 
>> believe this
>> >"will" strength is too weak. Think of a stateless proxy 
>that performs
>> >authentication. If it receives an ACK with no credentials, 
>> it knows it
>> >should not challenge the ACK. But what does it do? It has 
>two options
>> >and it will have a hard time figuring out when to do what:
>> >
>> >1. Drop it. But an ACK may be legitimately without 
>> credentials (like ACK
>> >for 200) and should be sent along.
>> >
>> >2. Proxy it. But the INVITE may have been challenged (ACK 
>> for 407) and
>> >the UAS will receive ACK out of the blue. Not detrimental 
>> but certainly
>> >not nice.
>> >
>> >So I think the "will"  requirement on UAC to include same 
>> Authorization
>> >header as INVITE in ACK needs to be strengthened to "MUST" level.
>> >
>> >Regards,
>> >
>> >Shan Lu
>> >
>> >sentitO Networks
>> >
>> >
>> >_______________________________________________
>> >Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> >This list is for NEW development of the core SIP Protocol
>> >Use sip-implementors@cs.columbia.edu for questions on current sip
>> >Use sipping@ietf.org for new developments on the application of sip
>> 
>> 
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>> 
>> 
>> 
>> 
>> 
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>> 
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip
>


_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From daemon@optimus.ietf.org  Thu Jul 25 10:20:42 2002
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From: "Sp.Raja" <spraja@isofttechindia.com>
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Subject: [Sip] Sec 28.1 clarification
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Sec 28.1 of RFC 3261 has a bullet which reads

o Similarly, CANCEL and BYE transactions were intermingled; RFC 2543
allowed the UAS not to send a response to INVITE when a BYE was
received. That is disallowed here. The original INVITE needs a
response.

But I suppose it should be

o Similarly, *INVITE* and BYE transactions were intermingled; RFC 2543
allowed the UAS not to send a response to INVITE when a BYE was
received. That is disallowed here. The original INVITE needs a
response.

Correct me if I'm wrong.

-Sp.Raja

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From daemon@optimus.ietf.org  Thu Jul 25 12:19:22 2002
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From: Henning Schulzrinne <hgs@cs.columbia.edu>
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Subject: Re: [Sip] language info in SIP headers
References: <3D3CE1C9.4080505@jp.fujitsu.com> <3D3D0C1A.7020307@cs.columbia.edu> <3D3D22C6.2080208@jp.fujitsu.com> <3D3D268B.8030803@cs.columbia.edu> <3D3F4EA8.6090501@jp.fujitsu.com>
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The chairs will probably be helpful to provide guidance; I would imagine 
the formal approach would be a separate requirements document. Another 
pragmatic approach might be a document that presents and motivates the 
problem to be solved and then presents some solutions and why they may 
or may not work.

Shingo Fujimoto wrote:
> Henning Schulzrinne wrote:

> Yes, I will.
> 
> 
> I just wonder if I should write it as a requirement doc, or just a 
> specification proposal doc.
> 
> Because I think the requirements are some kind of simple, e.g. UTF-8 
> friendly and multiple language in same field.
> 


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This list is for NEW development of the core SIP Protocol
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From daemon@optimus.ietf.org  Thu Jul 25 14:32:33 2002
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Henning Schulzrinne'" <hgs@cs.columbia.edu>,
        Shingo Fujimoto
	 <shingo_fujimoto@jp.fujitsu.com>
Cc: SIP-WG <sip@ietf.org>
Subject: RE: [Sip] language info in SIP headers
Date: Thu, 25 Jul 2002 13:53:12 -0400
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I agree that it's requirements first.
I think if requirements fit in less than a page, than one ID with
requirements up front and possible solutions should be written up
in an I-D and presented to SIPPING.

Brian

> -----Original Message-----
> From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
> Sent: Thursday, July 25, 2002 11:42 AM
> To: Shingo Fujimoto
> Cc: SIP-WG
> Subject: Re: [Sip] language info in SIP headers
> 
> 
> The chairs will probably be helpful to provide guidance; I 
> would imagine 
> the formal approach would be a separate requirements 
> document. Another 
> pragmatic approach might be a document that presents and 
> motivates the 
> problem to be solved and then presents some solutions and why 
> they may 
> or may not work.
> 
> Shingo Fujimoto wrote:
> > Henning Schulzrinne wrote:
> 
> > Yes, I will.
> > 
> > 
> > I just wonder if I should write it as a requirement doc, or just a 
> > specification proposal doc.
> > 
> > Because I think the requirements are some kind of simple, 
> e.g. UTF-8 
> > friendly and multiple language in same field.
> > 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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From daemon@ns.ietf.org  Thu Jul 25 21:50:49 2002
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From: "Sriram Parameswar" <sriramp@nortelnetworks.com>
To: "'BECKMANN MARK'" <mark.beckmann@sal.siemens.de>,
        "'sip@ietf.org'"
	 <sip@ietf.org>
Subject: RE: [Sip] Comments on draft-rosenberg-sip-reg-00
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Hi,

I have a comment and a question.

Comment: Mark - I think there is a fairly big difference between "expired"
and "deactivated". When you simply do not re-register your device, your
registration goes to state Terminated with a reason of "expired" and when an
administrator specifically removes your binding, you go to state Terminated
with a reason of "deactivated". Let us suppose I have several devices and
have SUBSCRIBEd to their reg event, I would be very interested in finding
the difference between a device whose Registration expired due my inactivity
and one whose Registration expired due to a deliberate action on the part of
the administrator. It does not matter that in both cases I am allowed to
re-register.

Question: Jonathan/Mark - lets use the example that I have several devices
attached to an AoR of 'skp@example.com'.  Their contacts are
'skp@pc.example.com', 'skp@sip_phone.example.com' and
'skp@wireless_dev.example.com'. Now I want to authorize somebody to be able
to see my registration state on the 'PC' and 'SIP_phone' but not the
'Wireless_device' how do we achieve this? I think there are some privacy
concerns here that we may have skipped over, the fact that some Contact
address may be private and I do not want to share information about that,
while others are public.

Thanks,

Sriram

__________________________________________
Sriram Parameswar              Phone: 972-685-8540
Interactive Multimedia Server (IMS) Fax: 972-684-3986
Nortel Networks, Richardson USA  Email: sriramp@nortelnetworks.com


-----Original Message-----
From: BECKMANN MARK [mailto:mark.beckmann@sal.siemens.de]
Sent: Thursday, July 25, 2002 7:43 AM
To: 'sip@ietf.org'
Subject: [Sip] Comments on draft-rosenberg-sip-reg-00
Importance: High


Hi Jonathan,

in your draft on the event package for registrations, you proposed the
following FSM for the contacts registered with an adress of record:
                                                                        
                                                                          
                                 +------+                                 
                                 |      | refreshed                       
                                 |      |                                 
                                 V      |                                 
    +------------+            +------------+            +------------+    
    |            |            |            |            |            |    
    |    Init    |----------->|   Active   |----------->| Terminated |    
    |            |            |            |            |            |    
    +------------+ registered +------------+ expired    +------------+    
                   created                   deactivated                  
                                             probation                    
                                             unregistered                 
                                             rejected                     
                                                                          

In section 3.1 it is described that in case an administrator wants a device
to re-authenticate, the registration is terminated and the device is
notified, e.g. by setting the event parameter to "deactivated".

First of all, I can't see a big difference between the two events "expired"
and "deactivated". "Deactivated" seems to be part of the "expired" event. In
both cases the contact may be re-registered. It doesn't seem very important
whether the registration is moved to the terminated state because it has
expired or because it has been deactivated.

Secondly, I think in case an administrator wants a device to
re-authenticate, the registration of the corresponding contact should not
move to the terminated state. Instead, another event (for example
"re-authenticate") similar to the "refresh" event should be introduced. When
this event occurs, the device is notified and the duration for which the
contact remains registered is reduced and if the device fails to
re-register, the registration is moved to the terminated state e.g. via the
"expired" event.

Thus, the FSM would look like:
                                                                          
                                 +------+                                 
                                 |      | refreshed                       
                                 |      | re-authenticate

                                 V      |                                 
    +------------+            +------------+            +------------+    
    |            |            |            |            |            |    
    |    Init    |----------->|   Active   |----------->| Terminated |    
    |            |            |            |            |            |    
    +------------+ registered +------------+            +------------+    
                   created                   expired               
                                             probation            
                                             unregistered           
                                             rejected               

Best regards,


Mark


Mark Beckmann			Siemens AG

ICM MP P PS 4S2
P.O.Box 100702			phone: +49 (5341) 906 1814
D-38228 Salzgitter   		fax:   +49 (5341) 906 2010

mailto: Mark.Beckmann@siemens.com



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip

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<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
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<TITLE>RE: [Sip] Comments on draft-rosenberg-sip-reg-00</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>Hi,</FONT>
</P>

<P><FONT SIZE=3D2>I have a comment and a question.</FONT>
</P>

<P><FONT SIZE=3D2>Comment: Mark - I think there is a fairly big =
difference between &quot;expired&quot; and &quot;deactivated&quot;. =
When you simply do not re-register your device, your registration goes =
to state Terminated with a reason of &quot;expired&quot; and when an =
administrator specifically removes your binding, you go to state =
Terminated with a reason of &quot;deactivated&quot;. Let us suppose I =
have several devices and have SUBSCRIBEd to their reg event, I would be =
very interested in finding the difference between a device whose =
Registration expired due my inactivity and one whose Registration =
expired due to a deliberate action on the part of the administrator. It =
does not matter that in both cases I am allowed to =
re-register.</FONT></P>

<P><FONT SIZE=3D2>Question: Jonathan/Mark - lets use the example that I =
have several devices attached to an AoR of 'skp@example.com'.&nbsp; =
Their contacts are 'skp@pc.example.com', 'skp@sip_phone.example.com' =
and 'skp@wireless_dev.example.com'. Now I want to authorize somebody to =
be able to see my registration state on the 'PC' and 'SIP_phone' but =
not the 'Wireless_device' how do we achieve this? I think there are =
some privacy concerns here that we may have skipped over, the fact that =
some Contact address may be private and I do not want to share =
information about that, while others are public.</FONT></P>

<P><FONT SIZE=3D2>Thanks,</FONT>
</P>

<P><FONT SIZE=3D2>Sriram</FONT>
</P>

<P><FONT SIZE=3D2>__________________________________________</FONT>
<BR><FONT SIZE=3D2>Sriram =
Parameswar&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp; Phone: 972-685-8540</FONT>
<BR><FONT SIZE=3D2>Interactive Multimedia Server (IMS) Fax: =
972-684-3986</FONT>
<BR><FONT SIZE=3D2>Nortel Networks, Richardson USA&nbsp; Email: =
sriramp@nortelnetworks.com</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: BECKMANN MARK [<A =
HREF=3D"mailto:mark.beckmann@sal.siemens.de">mailto:mark.beckmann@sal.si=
emens.de</A>]</FONT>
<BR><FONT SIZE=3D2>Sent: Thursday, July 25, 2002 7:43 AM</FONT>
<BR><FONT SIZE=3D2>To: 'sip@ietf.org'</FONT>
<BR><FONT SIZE=3D2>Subject: [Sip] Comments on =
draft-rosenberg-sip-reg-00</FONT>
<BR><FONT SIZE=3D2>Importance: High</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Hi Jonathan,</FONT>
</P>

<P><FONT SIZE=3D2>in your draft on the event package for registrations, =
you proposed the</FONT>
<BR><FONT SIZE=3D2>following FSM for the contacts registered with an =
adress of record:</FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp; </FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp; </FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
+------+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | =
refreshed&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
V&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp; =
+------------+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp; =
+------------+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp; +------------+&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp; =
Init&nbsp;&nbsp;&nbsp; |-----------&gt;|&nbsp;&nbsp; Active&nbsp;&nbsp; =
|-----------&gt;| Terminated |&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp; +------------+ registered =
+------------+ expired&nbsp;&nbsp;&nbsp; =
+------------+&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
created&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
deactivated&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
probation&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
unregistered&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
rejected&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp; </FONT>
</P>

<P><FONT SIZE=3D2>In section 3.1 it is described that in case an =
administrator wants a device</FONT>
<BR><FONT SIZE=3D2>to re-authenticate, the registration is terminated =
and the device is</FONT>
<BR><FONT SIZE=3D2>notified, e.g. by setting the event parameter to =
&quot;deactivated&quot;.</FONT>
</P>

<P><FONT SIZE=3D2>First of all, I can't see a big difference between =
the two events &quot;expired&quot;</FONT>
<BR><FONT SIZE=3D2>and &quot;deactivated&quot;. &quot;Deactivated&quot; =
seems to be part of the &quot;expired&quot; event. In</FONT>
<BR><FONT SIZE=3D2>both cases the contact may be re-registered. It =
doesn't seem very important</FONT>
<BR><FONT SIZE=3D2>whether the registration is moved to the terminated =
state because it has</FONT>
<BR><FONT SIZE=3D2>expired or because it has been deactivated.</FONT>
</P>

<P><FONT SIZE=3D2>Secondly, I think in case an administrator wants a =
device to</FONT>
<BR><FONT SIZE=3D2>re-authenticate, the registration of the =
corresponding contact should not</FONT>
<BR><FONT SIZE=3D2>move to the terminated state. Instead, another event =
(for example</FONT>
<BR><FONT SIZE=3D2>&quot;re-authenticate&quot;) similar to the =
&quot;refresh&quot; event should be introduced. When</FONT>
<BR><FONT SIZE=3D2>this event occurs, the device is notified and the =
duration for which the</FONT>
<BR><FONT SIZE=3D2>contact remains registered is reduced and if the =
device fails to</FONT>
<BR><FONT SIZE=3D2>re-register, the registration is moved to the =
terminated state e.g. via the</FONT>
<BR><FONT SIZE=3D2>&quot;expired&quot; event.</FONT>
</P>

<P><FONT SIZE=3D2>Thus, the FSM would look like:</FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp; </FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
+------+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | =
refreshed&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | re-authenticate</FONT>
</P>

<P><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
V&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp; =
+------------+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp; =
+------------+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp; +------------+&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp; =
Init&nbsp;&nbsp;&nbsp; |-----------&gt;|&nbsp;&nbsp; Active&nbsp;&nbsp; =
|-----------&gt;| Terminated |&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp; +------------+ registered =
+------------+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp; +------------+&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
created&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
expired&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp; </FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
probation&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp; </FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
unregistered&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
 </FONT>
<BR><FONT =
SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
rejected&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp; </FONT>
</P>

<P><FONT SIZE=3D2>Best regards,</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Mark</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Mark Beckmann&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Siemens AG</FONT>
</P>

<P><FONT SIZE=3D2>ICM MP P PS 4S2</FONT>
<BR><FONT SIZE=3D2>P.O.Box 100702&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; phone: +49 (5341) 906 =
1814</FONT>
<BR><FONT SIZE=3D2>D-38228 Salzgitter&nbsp;&nbsp; &nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; fax:&nbsp;&nbsp; +49 (5341) =
906 2010</FONT>
</P>

<P><FONT SIZE=3D2>mailto: Mark.Beckmann@siemens.com</FONT>
</P>
<BR>
<BR>

<P><FONT SIZE=3D2>_______________________________________________</FONT>=

<BR><FONT SIZE=3D2>Sip mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sip" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sip</A></FONT>
<BR><FONT SIZE=3D2>This list is for NEW development of the core SIP =
Protocol</FONT>
<BR><FONT SIZE=3D2>Use sip-implementors@cs.columbia.edu for questions =
on current sip</FONT>
<BR><FONT SIZE=3D2>Use sipping@ietf.org for new developments on the =
application of sip</FONT>
</P>

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Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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Sir,


In 2.4.5 § of refer-06 draft It is stated that a "A minimal, but complete, 
implementation can respond with a single NOTIFY containing either ...the body: 
SIP/2.0 200 OK if the reference was successful".

It seems to be in contradiction with § 3.10 saying that "Alternatively, the 
subscription might only result in two NOTIFY requests, the immediate NOTIFY and 
the NOTIFY carrying the final result of the reference".

Do we need 1 or 2 NOTIFY in a minimal implementation?

Best Regards,

Patrick

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Fri Jul 26 08:04:59 2002
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It can be either 1 or more, but at least 1.
Qouting Section 2.4.7: "Each NOTIFY must contain a Subscription-State header field as defined
   in [2].  The final NOTIFY sent in response to a REFER MUST indicate
   the subscription has been "terminated" with a reason of "noresource"."

The final NOTIFY can also be the first NOTIFY.

Regards... Vijay

-----Original Message-----
From: patrick.mourot@online.fr [mailto:patrick.mourot@online.fr] 
Sent: Friday, July 26, 2002 2:48 PM
To: sip@ietf.org
Subject: [Sip] refer-06 question


Sir,


In 2.4.5 § of refer-06 draft It is stated that a "A minimal, but complete, 
implementation can respond with a single NOTIFY containing either ...the body: 
SIP/2.0 200 OK if the reference was successful".

It seems to be in contradiction with § 3.10 saying that "Alternatively, the 
subscription might only result in two NOTIFY requests, the immediate NOTIFY and 
the NOTIFY carrying the final result of the reference".

Do we need 1 or 2 NOTIFY in a minimal implementation?

Best Regards,

Patrick

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip Use sipping@ietf.org for new developments on the application of sip

_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Fri Jul 26 08:41:50 2002
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Subject: RE: [Sip] refer-06 question
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Date: Fri, 26 Jul 2002 13:53:50 +0200 (MEST)
From: patrick.mourot@online.fr
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vsk,

Thanks for the answer.
I understand both cases; but this does not light my way.
Let me put it in other words.

can i delay the notification until the referenced request completes?
leading to send only 1 notification???? (assuming timers are OK).

Or am i oblige to send a 1rst notification, wait for completion of the 
referenced request and send a final notification ?

What is mandatory?

Patrick


Quoting Vijay S Kamath <vijay.kamath@wipro.com>:

> It can be either 1 or more, but at least 1.
> Qouting Section 2.4.7: "Each NOTIFY must contain a Subscription-State
> header field as defined
>    in [2].  The final NOTIFY sent in response to a REFER MUST indicate
>    the subscription has been "terminated" with a reason of
> "noresource"."
> 
> The final NOTIFY can also be the first NOTIFY.
> 
> Regards... Vijay
> 
> -----Original Message-----
> From: patrick.mourot@online.fr [mailto:patrick.mourot@online.fr] 
> Sent: Friday, July 26, 2002 2:48 PM
> To: sip@ietf.org
> Subject: [Sip] refer-06 question
> 
> 
> Sir,
> 
> 
> In 2.4.5 § of refer-06 draft It is stated that a "A minimal, but
> complete, 
> implementation can respond with a single NOTIFY containing either ...the
> body: 
> SIP/2.0 200 OK if the reference was successful".
> 
> It seems to be in contradiction with § 3.10 saying that "Alternatively,
> the 
> subscription might only result in two NOTIFY requests, the immediate
> NOTIFY and 
> the NOTIFY carrying the final result of the reference".
> 
> Do we need 1 or 2 NOTIFY in a minimal implementation?
> 
> Best Regards,
> 
> Patrick
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip Use
> sipping@ietf.org for new developments on the application of sip
> 

_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From daemon@optimus.ietf.org  Fri Jul 26 09:00:09 2002
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--NextPart

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

	Title		: Session Initiation Protocol Extension for Instant 
                          Messaging
	Author(s)	: B. Campbell, J. Rosenberg
	Filename	: draft-ietf-sip-message-06.txt
	Pages		: 22
	Date		: 25-Jul-02
	
Instant Messageing (IM) refers to the transfer of messages between
users in near real-time.  These messages are usually, but not
required to be, short.  IMs are often used in a conversational mode,
that is, the transfer of messages back and forth is fast enough for
participants to maintain an interactive conversation.
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(SIP) that allows the transfer of Instant Messages.  MESSAGE requests
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From daemon@optimus.ietf.org  Fri Jul 26 09:01:25 2002
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From: "Vijay S Kamath" <vijay.kamath@wipro.com>
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Patrick,

You cannot delay the notification until the referenced request completes.
Quoting Sec 2.4.4: "The creation of a subscription as defined by [1] always results in an
immediate NOTIFY. "

So I suggest you follow your second suggestion.

Vijay

-----Original Message-----
From: patrick.mourot@online.fr [mailto:patrick.mourot@online.fr] 
Sent: Friday, July 26, 2002 5:24 PM
To: Vijay Kamath
Cc: sip@ietf.org
Subject: RE: [Sip] refer-06 question


vsk,

Thanks for the answer.
I understand both cases; but this does not light my way.
Let me put it in other words.

can i delay the notification until the referenced request completes? leading to send only 1 notification???? (assuming timers are OK).

Or am i oblige to send a 1rst notification, wait for completion of the 
referenced request and send a final notification ?

What is mandatory?

Patrick


Quoting Vijay S Kamath <vijay.kamath@wipro.com>:

> It can be either 1 or more, but at least 1.
> Qouting Section 2.4.7: "Each NOTIFY must contain a Subscription-State 
> header field as defined
>    in [2].  The final NOTIFY sent in response to a REFER MUST indicate
>    the subscription has been "terminated" with a reason of 
> "noresource"."
> 
> The final NOTIFY can also be the first NOTIFY.
> 
> Regards... Vijay
> 
> -----Original Message-----
> From: patrick.mourot@online.fr [mailto:patrick.mourot@online.fr]
> Sent: Friday, July 26, 2002 2:48 PM
> To: sip@ietf.org
> Subject: [Sip] refer-06 question
> 
> 
> Sir,
> 
> 
> In 2.4.5 § of refer-06 draft It is stated that a "A minimal, but 
> complete, implementation can respond with a single NOTIFY containing 
> either ...the
> body:
> SIP/2.0 200 OK if the reference was successful".
> 
> It seems to be in contradiction with § 3.10 saying that 
> "Alternatively, the subscription might only result in two NOTIFY 
> requests, the immediate NOTIFY and
> the NOTIFY carrying the final result of the reference".
> 
> Do we need 1 or 2 NOTIFY in a minimal implementation?
> 
> Best Regards,
> 
> Patrick
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip Use 
> sipping@ietf.org for new developments on the application of sip
> 

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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Hi Sriram,
 
please see my comments inline.
 
 
Best regards,
 
Mark
 

Mark Beckmann                   Siemens AG 

ICM MP P PS 4S2 
P.O.Box 100702                  phone: +49 (5341) 906 1814 
D-38228 Salzgitter              fax:   +49 (5341) 906 2010 

mailto: Mark.Beckmann@siemens.com 

-----Original Message-----
From: Sriram Parameswar [mailto:sriramp@nortelnetworks.com]
Sent: Friday, July 26, 2002 3:04 AM
To: BECKMANN MARK; 'sip@ietf.org'
Subject: RE: [Sip] Comments on draft-rosenberg-sip-reg-00



Hi, 

I have a comment and a question. 

Comment: Mark - I think there is a fairly big difference between "expired"
and "deactivated". When you simply do not re-register your device, your
registration goes to state Terminated with a reason of "expired" and when an
administrator specifically removes your binding, you go to state Terminated
with a reason of "deactivated". Let us suppose I have several devices and
have SUBSCRIBEd to their reg event, I would be very interested in finding
the difference between a device whose Registration expired due my inactivity
and one whose Registration expired due to a deliberate action on the part of
the administrator. It does not matter that in both cases I am allowed to
re-register.
 
[MB] Agreed. My  initial thought was that "deactivated" could be seen as an
"expired" with the only difference that the administrator has reduced the
duration for which the registration is valid without explicitely telling the
device.

Question: Jonathan/Mark - lets use the example that I have several devices
attached to an AoR of 'skp@example.com'.  Their contacts are
'skp@pc.example.com', 'skp@sip_phone.example.com' and
'skp@wireless_dev.example.com'. Now I want to authorize somebody to be able
to see my registration state on the 'PC' and 'SIP_phone' but not the
'Wireless_device' how do we achieve this? I think there are some privacy
concerns here that we may have skipped over, the fact that some Contact
address may be private and I do not want to share information about that,
while others are public.
 
[MB]  I agree that there is a privacy issue here, which is probably very
similar to the one for presence. Therefore it should be stated that a
subscription may be authenticated and authorized, whereby the mechanism for
authorization is outside the scope of the reg-event draft. The watcher info
package may be used for the reg event to inform the "owner" of the
registration about subscriptions.

   Thanks, 

Sriram 

__________________________________________ 
Sriram Parameswar              Phone: 972-685-8540 
Interactive Multimedia Server (IMS) Fax: 972-684-3986 
Nortel Networks, Richardson USA  Email: sriramp@nortelnetworks.com 


-----Original Message----- 
From: BECKMANN MARK [ mailto:mark.beckmann@sal.siemens.de
<mailto:mark.beckmann@sal.siemens.de> ] 
Sent: Thursday, July 25, 2002 7:43 AM 
To: 'sip@ietf.org' 
Subject: [Sip] Comments on draft-rosenberg-sip-reg-00 
Importance: High 


Hi Jonathan, 

in your draft on the event package for registrations, you proposed the 
following FSM for the contacts registered with an adress of record: 
                                                                        
                                                                          
                                 +------+                                 
                                 |      | refreshed                       
                                 |      |                                 
                                 V      |                                 
    +------------+            +------------+            +------------+    
    |            |            |            |            |            |    
    |    Init    |----------->|   Active   |----------->| Terminated |    
    |            |            |            |            |            |    
    +------------+ registered +------------+ expired    +------------+    
                   created                   deactivated                  
                                             probation                    
                                             unregistered                 
                                             rejected                     
                                                                          

In section 3.1 it is described that in case an administrator wants a device 
to re-authenticate, the registration is terminated and the device is 
notified, e.g. by setting the event parameter to "deactivated". 

First of all, I can't see a big difference between the two events "expired" 
and "deactivated". "Deactivated" seems to be part of the "expired" event. In

both cases the contact may be re-registered. It doesn't seem very important 
whether the registration is moved to the terminated state because it has 
expired or because it has been deactivated. 

Secondly, I think in case an administrator wants a device to 
re-authenticate, the registration of the corresponding contact should not 
move to the terminated state. Instead, another event (for example 
"re-authenticate") similar to the "refresh" event should be introduced. When

this event occurs, the device is notified and the duration for which the 
contact remains registered is reduced and if the device fails to 
re-register, the registration is moved to the terminated state e.g. via the 
"expired" event. 

Thus, the FSM would look like: 
                                                                          
                                 +------+                                 
                                 |      | refreshed                       
                                 |      | re-authenticate 

                                 V      |                                 
    +------------+            +------------+            +------------+    
    |            |            |            |            |            |    
    |    Init    |----------->|   Active   |----------->| Terminated |    
    |            |            |            |            |            |    
    +------------+ registered +------------+            +------------+    
                   created                   expired               
                                             probation            
                                             unregistered           
                                             rejected               

Best regards, 


Mark 


Mark Beckmann                   Siemens AG 

ICM MP P PS 4S2 
P.O.Box 100702                  phone: +49 (5341) 906 1814 
D-38228 Salzgitter              fax:   +49 (5341) 906 2010 

mailto: Mark.Beckmann@siemens.com 



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<TITLE>RE: [Sip] Comments on draft-rosenberg-sip-reg-00</TITLE>

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<DIV><SPAN class=697212012-26072002><FONT face=Arial color=#0000ff size=2>Hi 
Sriram,</FONT></SPAN></DIV>
<DIV><SPAN class=697212012-26072002><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=697212012-26072002><FONT face=Arial color=#0000ff size=2>please 
see my comments inline.</FONT></SPAN></DIV>
<DIV><SPAN class=697212012-26072002><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=697212012-26072002><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=697212012-26072002><FONT face=Arial color=#0000ff size=2>Best 
regards,</FONT></SPAN></DIV>
<DIV><SPAN class=697212012-26072002><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=697212012-26072002><FONT face=Arial color=#0000ff 
size=2>Mark</FONT></SPAN></DIV>
<DIV>&nbsp;</DIV>
<P><FONT face=Arial size=1>Mark Beckmann&nbsp;&nbsp; 
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<B> </B></FONT><B><FONT face=Arial 
color=#008080 size=1>Siemens AG</FONT></B> </P>
<P><FONT face=Arial size=1>ICM MP P PS 4S2</FONT> <BR><FONT face=Arial 
size=1>P.O.Box 100702&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; phone:</FONT><FONT face=Courier 
size=1></FONT> <FONT face=Arial size=1>+49 (5341) 906 1814</FONT> <BR><FONT 
face=Arial size=1>D-38228 Salzgitter&nbsp;&nbsp; &nbsp;&nbsp; 
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; fax:</FONT><FONT face=Courier 
size=1>&nbsp;&nbsp;</FONT> <FONT face=Arial size=1>+49 (5341) 906 2010</FONT> 
</P>
<P><FONT face=Arial size=1>mailto: Mark.Beckmann@siemens.com</FONT> </P>
<BLOCKQUOTE dir=ltr 
style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px">
  <DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> Sriram Parameswar 
  [mailto:sriramp@nortelnetworks.com]<BR><B>Sent:</B> Friday, July 26, 2002 3:04 
  AM<BR><B>To:</B> BECKMANN MARK; 'sip@ietf.org'<BR><B>Subject:</B> RE: [Sip] 
  Comments on draft-rosenberg-sip-reg-00<BR><BR></FONT></DIV>
  <P><FONT size=2>Hi,</FONT> </P>
  <P><FONT size=2>I have a comment and a question.</FONT> </P>
  <P><FONT size=2>Comment: Mark - I think there is a fairly big difference 
  between "expired" and "deactivated". When you simply do not re-register your 
  device, your registration goes to state Terminated with a reason of "expired" 
  and when an administrator specifically removes your binding, you go to state 
  Terminated with a reason of "deactivated". Let us suppose I have several 
  devices and have SUBSCRIBEd to their reg event, I would be very interested in 
  finding the difference between a device whose Registration expired due my 
  inactivity and one whose Registration expired due to a deliberate action on 
  the part of the administrator. It does not matter that in both cases I am 
  allowed to re-register.<BR><SPAN class=697212012-26072002><FONT face=Arial 
  color=#0000ff>&nbsp;</FONT></SPAN><BR><SPAN class=697212012-26072002><FONT 
  face=Arial color=#0000ff>[MB]&nbsp;Agreed. My &nbsp;initial thought was that 
  "deactivated" could be seen as an "expired" with the only difference that the 
  administrator has reduced the duration for which the registration is valid 
  without explicitely telling the device.</FONT></SPAN></FONT></P>
  <P><FONT size=2>Question: Jonathan/Mark - lets use the example that I have 
  several devices attached to an AoR of 'skp@example.com'.&nbsp; Their contacts 
  are 'skp@pc.example.com', 'skp@sip_phone.example.com' and 
  'skp@wireless_dev.example.com'. Now I want to authorize somebody to be able to 
  see my registration state on the 'PC' and 'SIP_phone' but not the 
  'Wireless_device' how do we achieve this? I think there are some privacy 
  concerns here that we may have skipped over, the fact that some Contact 
  address may be private and I do not want to share information about that, 
  while others are public.<BR><SPAN class=697212012-26072002><FONT face=Arial 
  color=#0000ff>&nbsp;</FONT></SPAN><BR><SPAN class=697212012-26072002><FONT 
  face=Arial color=#0000ff>[MB]&nbsp;&nbsp;I agree that there is a privacy issue 
  here, which is probably very similar to the one for presence.&nbsp;Therefore 
  it should be stated that a subscription may be authenticated and authorized, 
  whereby the mechanism for authorization is outside the scope of the reg-event 
  draft. The watcher info package may be used for the reg event to inform the 
  "owner" of the registration about subscriptions.</FONT></SPAN></FONT></P>
  <P><FONT size=2><SPAN class=697212012-26072002>&nbsp;&nbsp; 
  </SPAN></FONT><FONT size=2>Thanks,</FONT> </P>
  <P><FONT size=2>Sriram</FONT> </P>
  <P><FONT size=2>__________________________________________</FONT> <BR><FONT 
  size=2>Sriram 
  Parameswar&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  Phone: 972-685-8540</FONT> <BR><FONT size=2>Interactive Multimedia Server 
  (IMS) Fax: 972-684-3986</FONT> <BR><FONT size=2>Nortel Networks, Richardson 
  USA&nbsp; Email: sriramp@nortelnetworks.com</FONT> </P><BR>
  <P><FONT size=2>-----Original Message-----</FONT> <BR><FONT size=2>From: 
  BECKMANN MARK [<A 
  href="mailto:mark.beckmann@sal.siemens.de">mailto:mark.beckmann@sal.siemens.de</A>]</FONT> 
  <BR><FONT size=2>Sent: Thursday, July 25, 2002 7:43 AM</FONT> <BR><FONT 
  size=2>To: 'sip@ietf.org'</FONT> <BR><FONT size=2>Subject: [Sip] Comments on 
  draft-rosenberg-sip-reg-00</FONT> <BR><FONT size=2>Importance: High</FONT> 
  </P><BR>
  <P><FONT size=2>Hi Jonathan,</FONT> </P>
  <P><FONT size=2>in your draft on the event package for registrations, you 
  proposed the</FONT> <BR><FONT size=2>following FSM for the contacts registered 
  with an adress of record:</FONT> <BR><FONT 
  size=2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  </FONT><BR><FONT 
  size=2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  </FONT><BR><FONT 
  size=2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  +------+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  </FONT><BR><FONT 
  size=2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | 
  refreshed&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  </FONT><BR><FONT 
  size=2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  </FONT><BR><FONT 
  size=2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  V&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  </FONT><BR><FONT size=2>&nbsp;&nbsp;&nbsp; 
  +------------+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  +------------+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  +------------+&nbsp;&nbsp;&nbsp; </FONT><BR><FONT size=2>&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp; </FONT><BR><FONT size=2>&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp; Init&nbsp;&nbsp;&nbsp; |-----------&gt;|&nbsp;&nbsp; 
  Active&nbsp;&nbsp; |-----------&gt;| Terminated |&nbsp;&nbsp;&nbsp; 
  </FONT><BR><FONT size=2>&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp; </FONT><BR><FONT size=2>&nbsp;&nbsp;&nbsp; +------------+ 
  registered +------------+ expired&nbsp;&nbsp;&nbsp; 
  +------------+&nbsp;&nbsp;&nbsp; </FONT><BR><FONT 
  size=2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  created&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  deactivated&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  </FONT><BR><FONT 
  size=2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  probation&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  </FONT><BR><FONT 
  size=2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  unregistered&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  </FONT><BR><FONT 
  size=2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  rejected&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  </FONT><BR><FONT 
  size=2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  </FONT></P>
  <P><FONT size=2>In section 3.1 it is described that in case an administrator 
  wants a device</FONT> <BR><FONT size=2>to re-authenticate, the registration is 
  terminated and the device is</FONT> <BR><FONT size=2>notified, e.g. by setting 
  the event parameter to "deactivated".</FONT> </P>
  <P><FONT size=2>First of all, I can't see a big difference between the two 
  events "expired"</FONT> <BR><FONT size=2>and "deactivated". "Deactivated" 
  seems to be part of the "expired" event. In</FONT> <BR><FONT size=2>both cases 
  the contact may be re-registered. It doesn't seem very important</FONT> 
  <BR><FONT size=2>whether the registration is moved to the terminated state 
  because it has</FONT> <BR><FONT size=2>expired or because it has been 
  deactivated.</FONT> </P>
  <P><FONT size=2>Secondly, I think in case an administrator wants a device 
  to</FONT> <BR><FONT size=2>re-authenticate, the registration of the 
  corresponding contact should not</FONT> <BR><FONT size=2>move to the 
  terminated state. Instead, another event (for example</FONT> <BR><FONT 
  size=2>"re-authenticate") similar to the "refresh" event should be introduced. 
  When</FONT> <BR><FONT size=2>this event occurs, the device is notified and the 
  duration for which the</FONT> <BR><FONT size=2>contact remains registered is 
  reduced and if the device fails to</FONT> <BR><FONT size=2>re-register, the 
  registration is moved to the terminated state e.g. via the</FONT> <BR><FONT 
  size=2>"expired" event.</FONT> </P>
  <P><FONT size=2>Thus, the FSM would look like:</FONT> <BR><FONT 
  size=2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  </FONT><BR><FONT 
  size=2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  +------+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  </FONT><BR><FONT 
  size=2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | 
  refreshed&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  </FONT><BR><FONT 
  size=2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; | re-authenticate</FONT> </P>
  <P><FONT 
  size=2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  V&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  </FONT><BR><FONT size=2>&nbsp;&nbsp;&nbsp; 
  +------------+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  +------------+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  +------------+&nbsp;&nbsp;&nbsp; </FONT><BR><FONT size=2>&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp; </FONT><BR><FONT size=2>&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp; Init&nbsp;&nbsp;&nbsp; |-----------&gt;|&nbsp;&nbsp; 
  Active&nbsp;&nbsp; |-----------&gt;| Terminated |&nbsp;&nbsp;&nbsp; 
  </FONT><BR><FONT size=2>&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  |&nbsp;&nbsp;&nbsp; </FONT><BR><FONT size=2>&nbsp;&nbsp;&nbsp; +------------+ 
  registered 
  +------------+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  +------------+&nbsp;&nbsp;&nbsp; </FONT><BR><FONT 
  size=2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  created&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  expired&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  </FONT><BR><FONT 
  size=2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  probation&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  </FONT><BR><FONT 
  size=2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  unregistered&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  </FONT><BR><FONT 
  size=2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  rejected&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  </FONT></P>
  <P><FONT size=2>Best regards,</FONT> </P><BR>
  <P><FONT size=2>Mark</FONT> </P><BR>
  <P><FONT size=2>Mark Beckmann&nbsp;&nbsp; 
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Siemens AG</FONT> </P>
  <P><FONT size=2>ICM MP P PS 4S2</FONT> <BR><FONT size=2>P.O.Box 100702&nbsp; 
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; phone: +49 (5341) 906 1814</FONT> 
  <BR><FONT size=2>D-38228 Salzgitter&nbsp;&nbsp; &nbsp;&nbsp; 
  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; fax:&nbsp;&nbsp; +49 (5341) 906 
  2010</FONT> </P>
  <P><FONT size=2>mailto: Mark.Beckmann@siemens.com</FONT> </P><BR><BR>
  <P><FONT size=2>_______________________________________________</FONT> 
  <BR><FONT size=2>Sip mailing list&nbsp; <A target=_blank 
  href="https://www1.ietf.org/mailman/listinfo/sip">https://www1.ietf.org/mailman/listinfo/sip</A></FONT> 
  <BR><FONT size=2>This list is for NEW development of the core SIP 
  Protocol</FONT> <BR><FONT size=2>Use sip-implementors@cs.columbia.edu for 
  questions on current sip</FONT> <BR><FONT size=2>Use sipping@ietf.org for new 
  developments on the application of sip</FONT> </P></BLOCKQUOTE></BODY></HTML>

------_=_NextPart_001_01C234A2.2D0D7C40--

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From daemon@optimus.ietf.org  Fri Jul 26 11:34:24 2002
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From: David Oliver <Oliverdc@securelogix.com>
To: "'Rosen, Brian'" <Brian.Rosen@marconi.com>,
        "'Eiji Tomimura'"
	 <tomimura@sei.co.jp>, sip@ietf.org
Subject: RE: [Sip] Question about ABNF for SIP
Date: Fri, 26 Jul 2002 10:01:29 -0500
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Eiji,

Although an implementation MUST generate precisely correct (RFC3261) syntax,
it SHOULD accept all syntax it can interpret (i.e. not object to technical
errors where the meaning is still clear), according to the "robustness
principle" defined in section 3.2 of RFC791 [4]:

    an implementation must be conservative in its sending
    behavior, and liberal in its receiving behavior.

This would allow (for example) your implementation to interwork with other
implementations based erroneously on the bis-09 syntax, manually entered
scripts, or with other minor bugs such as misunderstood printf formats.

For example, a more robust acceptance syntax could be:

 rfc1123-date = wkday *SP "," *SP date1 1*SP time 1*SP "GMT"
 date1        = 1*2DIGIT 1*SP month 1*SP 4DIGIT
 time         = 1*2DIGIT ":" *SP 1*2DIGIT ":" *SP 1*2DIGIT

This would accept the following dates:

 Fri, 05 Jul 2002 09:40:01 GMT     -- strict syntax.
 Fri,  5 Jul 2002  9:40: 1 GMT     -- printf with leading zeros as space.
 Fri, 5 Jul 2002 9:40:1 GMT	     -- printf with missing leading zeros.
 Fri ,  05 Jul 2002  09:40:01 GMT  -- hand entered, with bad spacing
                                   -- due to variable sized font.

The acceptance syntax is not significantly more difficult to implement
except in the smallest embedded implementations, and does not cause
ambiguity. Of course, if the extended syntax caused performance or other
problems, it would be the first candidate for removal.

Regards,

David C. Oliver
Principal Engineer
SecureLogix Corp
Email: doliver@securelogix.com <mailto:doliver@securelogix.com>
Tel:   210.402.9669
Fax:   210.402.6996 



-----Original Message-----
From: Rosen, Brian [mailto:Brian.Rosen@marconi.com]
Sent: Wednesday, July 03, 2002 7:00 AM
To: 'Eiji Tomimura'; sip@ietf.org
Subject: RE: [Sip] Question about ABNF for SIP


This problem has been reported before.  We fixed it in the
text of the forthcoming RFC.  The SP is added.

Brian

> -----Original Message-----
> From: Eiji Tomimura [mailto:tomimura@sei.co.jp]
> Sent: Wednesday, July 03, 2002 5:10 AM
> To: sip@ietf.org
> Subject: [Sip] Question about ABNF for SIP
> 
> 
> Dear all,
> 
> I have a question regarding ABNF for SIP.
> 
> In bis-09, it defines that :
>     rfc1123-date = wkday "," date1 SP time SP "GMT"
> 
> Though, in RFC2616 "HTTP/1.1", it defines that:
>     rfc1123-date = wkday "," SP date1 SP time SP "GMT"
> 
> Is it intentional? Or, is it a bug of either side?
> 
> Becauses our SIP layer is based on RFC2616 format, it reject 
> Date header
> follows the format of bis-09 as syntax error. So I'm 
> wondering which format
> should we use.
> 
> Thanks in advance.
> 
> Regards,
> - Eiji Tomimura
> 
> -- 
> Eiji Tomimura 
> Senior Engineer 
> Communications Technology Dept., Information Technology R&D 
> Laboratories
> Sumitomo Electric Industries, Ltd.
> Phone: +81-6-6466-5643  Fax: +81-6-6466-5732  Email: 
> tomimura@sei.co.jp
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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From: =?iso-8859-1?Q?Manuel_Moreno_Mart=EDn?= <mmoreno@cipres.upm.es>
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mmoreno@cipres.upm.es

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From daemon@optimus.ietf.org  Fri Jul 26 12:08:52 2002
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From: =?iso-8859-1?Q?Manuel_Moreno_Mart=EDn?= <mmoreno@cipres.upm.es>
To: <sip@ietf.org>
Date: Fri, 26 Jul 2002 17:37:32 +0200
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I want to know more over the body of SIP messages.

In the book "UNDERSTANDING THE SESSION INITIATION PROTOCOL", of Alan B.
Johnston (page 80), I have read : "For 200 OK responses to BYE, CANCEL,
INFO,  and PRACK, a message body is not permitted"

But now, in the RFC 3261 (SIP), page 33, I read: "All responses MAY include
a body"

Please, I want to know more over this issue

Thank you for your time

Manuel


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From daemon@optimus.ietf.org  Fri Jul 26 15:47:45 2002
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Subject: Re: [Sip] Message body for 200 response
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On Friday, July 26, 2002, at 08:37 AM, Manuel Moreno Martín wrote:

> I want to know more over the body of SIP messages.
>
> In the book "UNDERSTANDING THE SESSION INITIATION PROTOCOL", of Alan B.
> Johnston (page 80), I have read : "For 200 OK responses to BYE, CANCEL,
> INFO,  and PRACK, a message body is not permitted"
>
> But now, in the RFC 3261 (SIP), page 33, I read: "All responses MAY 
> include
> a body"

RFC 3261 takes precedence until another WG document updates or obsoletes 
it.

thx,
-


> Please, I want to know more over this issue
>
> Thank you for your time
>
> Manuel
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Subject: Re: [Sip] refer-06 question
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Cc: Vijay S Kamath <vijay.kamath@wipro.com>, sip@ietf.org
To: patrick.mourot@online.fr
From: Rohan Mahy <rohan@cisco.com>
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On Friday, July 26, 2002, at 04:53 AM, patrick.mourot@online.fr wrote:

> vsk,
>
> Thanks for the answer.
> I understand both cases; but this does not light my way.
> Let me put it in other words.
>
> can i delay the notification until the referenced request completes?
> leading to send only 1 notification???? (assuming timers are OK).

hi,

not really.  its almost impossible to be sure that you can make sure the 
timers are OK.

> Or am i oblige to send a 1rst notification, wait for completion of the
> referenced request and send a final notification ?

the later.

so a minimal implementation would send first an immediate NOTIFY with 
the following body

SIP/2.0 100 Trying

and then upon success/failure a 200, 501, etc.. as described in the 
draft now.

thanks,
-rohan

> What is mandatory?
>
> Patrick
>
>
> Quoting Vijay S Kamath <vijay.kamath@wipro.com>:
>
>> It can be either 1 or more, but at least 1.
>> Qouting Section 2.4.7: "Each NOTIFY must contain a Subscription-State
>> header field as defined
>>    in [2].  The final NOTIFY sent in response to a REFER MUST indicate
>>    the subscription has been "terminated" with a reason of
>> "noresource"."
>>
>> The final NOTIFY can also be the first NOTIFY.
>>
>> Regards... Vijay
>>
>> -----Original Message-----
>> From: patrick.mourot@online.fr [mailto:patrick.mourot@online.fr]
>> Sent: Friday, July 26, 2002 2:48 PM
>> To: sip@ietf.org
>> Subject: [Sip] refer-06 question
>>
>>
>> Sir,
>>
>>
>> In 2.4.5 § of refer-06 draft It is stated that a "A minimal, but
>> complete,
>> implementation can respond with a single NOTIFY containing either 
>> ...the
>> body:
>> SIP/2.0 200 OK if the reference was successful".
>>
>> It seems to be in contradiction with § 3.10 saying that "Alternatively,
>> the
>> subscription might only result in two NOTIFY requests, the immediate
>> NOTIFY and
>> the NOTIFY carrying the final result of the reference".
>>
>> Do we need 1 or 2 NOTIFY in a minimal implementation?
>>
>> Best Regards,
>>
>> Patrick
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip Use
>> sipping@ietf.org for new developments on the application of sip
>>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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From daemon@optimus.ietf.org  Sat Jul 27 12:40:07 2002
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Subject: Re: [Sip] refer-06 question
References: <60614B15188EBB4FA59F85310C2AC7D620AE88@blr-m1-msg.wipro.com>
        <1027684430.3d41384e6a93d@imp.free.fr>
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From: Pekka Pessi <Pekka.Pessi@nokia.com>
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"patrick.mourot@online.fr" <patrick.mourot@online.fr> writes:
>can i delay the notification until the referenced request completes?
>leading to send only 1 notification???? (assuming timers are OK).

	No. You have to send the initial notification request. The
	initial NOTIFY request is needed in order to establish the
	dialog. Also, if the party sending does not care about the
	eventual result, it can respond with an error message to the
	initial NOTIFY and get the notification dialog terminated.

>Or am i oblige to send a 1rst notification, wait for completion of the 
>referenced request and send a final notification ?

	That is the current interpretation, I think.

						Pekka

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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: An Extension to the Session Initiation protocol to 
                          Assure Congestion Safety
	Author(s)	: D. Willis, B. Campbell
	Filename	: draft-willis-sip-congestsafe-00.txt
	Pages		: 11
	Date		: 24-Jul-02
	
The Session Initiation Protocol allows the use of UDP for transport
of SIP messages.  The use of UDP inherently risks network congestion
problems, as UDP itself does not define congestion prevention,
avoidance, detection, or correction mechanisms.  This problem is
aggravated by large SIP messages which fragment at the UDP level.
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From daemon@optimus.ietf.org  Mon Jul 29 05:25:48 2002
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Subject: RE: [Sip] refer-06 question
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To: <vijay.kamath@wipro.com>, <patrick.mourot@online.fr>
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I guess it all depends on how you define "immediate" to mean. If its a matter of seconds, then I think its ok to send the one NOTIFY only.

I quote RFC3165

"Until the first NOTIFY message arrives, the subscriber
   should consider the state of the subscribed resource to be in a
   neutral state.  Documents which define new event packages MUST define
   this "neutral state" in such a way that makes sense for their
   application (see section 4.4.7.)."

Regards,
Hisham

> -----Original Message-----
> From: ext Vijay S Kamath [mailto:vijay.kamath@wipro.com]
> Sent: Friday, July 26, 2002 3:25 PM
> To: patrick.mourot@online.fr
> Cc: sip@ietf.org
> Subject: RE: [Sip] refer-06 question
> 
> 
> Patrick,
> 
> You cannot delay the notification until the referenced 
> request completes.
> Quoting Sec 2.4.4: "The creation of a subscription as defined 
> by [1] always results in an
> immediate NOTIFY. "
> 
> So I suggest you follow your second suggestion.
> 
> Vijay
> 
> -----Original Message-----
> From: patrick.mourot@online.fr [mailto:patrick.mourot@online.fr] 
> Sent: Friday, July 26, 2002 5:24 PM
> To: Vijay Kamath
> Cc: sip@ietf.org
> Subject: RE: [Sip] refer-06 question
> 
> 
> vsk,
> 
> Thanks for the answer.
> I understand both cases; but this does not light my way.
> Let me put it in other words.
> 
> can i delay the notification until the referenced request 
> completes? leading to send only 1 notification???? (assuming 
> timers are OK).
> 
> Or am i oblige to send a 1rst notification, wait for 
> completion of the 
> referenced request and send a final notification ?
> 
> What is mandatory?
> 
> Patrick
> 
> 
> Quoting Vijay S Kamath <vijay.kamath@wipro.com>:
> 
> > It can be either 1 or more, but at least 1.
> > Qouting Section 2.4.7: "Each NOTIFY must contain a 
> Subscription-State 
> > header field as defined
> >    in [2].  The final NOTIFY sent in response to a REFER 
> MUST indicate
> >    the subscription has been "terminated" with a reason of 
> > "noresource"."
> > 
> > The final NOTIFY can also be the first NOTIFY.
> > 
> > Regards... Vijay
> > 
> > -----Original Message-----
> > From: patrick.mourot@online.fr [mailto:patrick.mourot@online.fr]
> > Sent: Friday, July 26, 2002 2:48 PM
> > To: sip@ietf.org
> > Subject: [Sip] refer-06 question
> > 
> > 
> > Sir,
> > 
> > 
> > In 2.4.5 § of refer-06 draft It is stated that a "A minimal, but 
> > complete, implementation can respond with a single NOTIFY 
> containing 
> > either ...the
> > body:
> > SIP/2.0 200 OK if the reference was successful".
> > 
> > It seems to be in contradiction with § 3.10 saying that 
> > "Alternatively, the subscription might only result in two NOTIFY 
> > requests, the immediate NOTIFY and
> > the NOTIFY carrying the final result of the reference".
> > 
> > Do we need 1 or 2 NOTIFY in a minimal implementation?
> > 
> > Best Regards,
> > 
> > Patrick
> > 
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on 
> current sip Use 
> > sipping@ietf.org for new developments on the application of sip
> > 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

_______________________________________________
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From daemon@optimus.ietf.org  Mon Jul 29 05:55:03 2002
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To: hisham.khartabil@nokia.com
Subject: RE: [Sip] refer-06 question
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Date: Mon, 29 Jul 2002 11:35:07 +0200 (MEST)
From: patrick.mourot@online.fr
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Assuming that you mean RFC3265, there are several contradictory answers.
I would like to suggest that refer-06 should be clearer on the subject.
It should describe what a minimal implementation is.

Patrick

Quoting hisham.khartabil@nokia.com:

> I guess it all depends on how you define "immediate" to mean. If its a
> matter of seconds, then I think its ok to send the one NOTIFY only.
> 
> I quote RFC3165
> 
> "Until the first NOTIFY message arrives, the subscriber
>    should consider the state of the subscribed resource to be in a
>    neutral state.  Documents which define new event packages MUST
> define
>    this "neutral state" in such a way that makes sense for their
>    application (see section 4.4.7.)."
> 
> Regards,
> Hisham
> 
> > -----Original Message-----
> > From: ext Vijay S Kamath [mailto:vijay.kamath@wipro.com]
> > Sent: Friday, July 26, 2002 3:25 PM
> > To: patrick.mourot@online.fr
> > Cc: sip@ietf.org
> > Subject: RE: [Sip] refer-06 question
> > 
> > 
> > Patrick,
> > 
> > You cannot delay the notification until the referenced 
> > request completes.
> > Quoting Sec 2.4.4: "The creation of a subscription as defined 
> > by [1] always results in an
> > immediate NOTIFY. "
> > 
> > So I suggest you follow your second suggestion.
> > 
> > Vijay
> > 
> > -----Original Message-----
> > From: patrick.mourot@online.fr [mailto:patrick.mourot@online.fr] 
> > Sent: Friday, July 26, 2002 5:24 PM
> > To: Vijay Kamath
> > Cc: sip@ietf.org
> > Subject: RE: [Sip] refer-06 question
> > 
> > 
> > vsk,
> > 
> > Thanks for the answer.
> > I understand both cases; but this does not light my way.
> > Let me put it in other words.
> > 
> > can i delay the notification until the referenced request 
> > completes? leading to send only 1 notification???? (assuming 
> > timers are OK).
> > 
> > Or am i oblige to send a 1rst notification, wait for 
> > completion of the 
> > referenced request and send a final notification ?
> > 
> > What is mandatory?
> > 
> > Patrick
> > 
> > 
> > Quoting Vijay S Kamath <vijay.kamath@wipro.com>:
> > 
> > > It can be either 1 or more, but at least 1.
> > > Qouting Section 2.4.7: "Each NOTIFY must contain a 
> > Subscription-State 
> > > header field as defined
> > >    in [2].  The final NOTIFY sent in response to a REFER 
> > MUST indicate
> > >    the subscription has been "terminated" with a reason of 
> > > "noresource"."
> > > 
> > > The final NOTIFY can also be the first NOTIFY.
> > > 
> > > Regards... Vijay
> > > 
> > > -----Original Message-----
> > > From: patrick.mourot@online.fr [mailto:patrick.mourot@online.fr]
> > > Sent: Friday, July 26, 2002 2:48 PM
> > > To: sip@ietf.org
> > > Subject: [Sip] refer-06 question
> > > 
> > > 
> > > Sir,
> > > 
> > > 
> > > In 2.4.5 § of refer-06 draft It is stated that a "A minimal, but 
> > > complete, implementation can respond with a single NOTIFY 
> > containing 
> > > either ...the
> > > body:
> > > SIP/2.0 200 OK if the reference was successful".
> > > 
> > > It seems to be in contradiction with § 3.10 saying that 
> > > "Alternatively, the subscription might only result in two NOTIFY 
> > > requests, the immediate NOTIFY and
> > > the NOTIFY carrying the final result of the reference".
> > > 
> > > Do we need 1 or 2 NOTIFY in a minimal implementation?
> > > 
> > > Best Regards,
> > > 
> > > Patrick
> > > 
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on 
> > current sip Use 
> > > sipping@ietf.org for new developments on the application of sip
> > > 
> > 
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> > 
> 

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From daemon@optimus.ietf.org  Mon Jul 29 06:06:19 2002
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Hi,

In section 4.1 Mapping of SIP Transactions into Streams,
it says
"..a SIP entity SHOULD send every SIP message
   (request or response) over stream zero with the unordered flag set..."

Are there no SIP extensions which may require ordered delivery  from
SCTP(sip-isup mapping for instance)?
or is the 'SHOULD'  supposed to accommodate such exceptions?

Later in the same section, it is said "..Some applications introduce an
extra layer between the transport
   layer and SIP (e.g., compression and/or encryption). This extra layer
   sometimes requires ordered delivery of messages from the transport
   layer (e.g., TLS [6])."

The section then recommends using same stream for maintaining
sequence...but no mention is made about whether the unordered flag should
be set or not. To maintain sequence, in addition to choosing the same
stream, will it not be required that messages be also sent in sequence
within a stream(by not setting the unordered flag)?


regards
Harsh Bhondwe
Hughes Software Systems






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Khartabil Hisham (NMP/Helsinki) would like to recall the message, "[Sip-implementors] Changes from bis-09 to RFC 3261".

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I trimmed those for our needs. I'll send the raw changes as sent to the sip mailing list in another following mail.

/Hisham

> -----Original Message-----
> From: ext Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Wednesday, July 24, 2002 8:52 AM
> To: sip@ietf.org; sip-implementors@cs.columbia.edu
> Subject: [Sip-implementors] Changes from bis-09 to RFC 3261
> 
> * It was reported that the processing of timer D in the transaction
> layer was wrong. This is the timer that fires in the INVITE client
> transaction after it has ceased waiting for retransmits of the final
> response, in order to send an ACK. Bis-09 said to inform the TU of a
> timeout. However, this is not a failure condition, so that makes no
> sense. As such, Section 17.1.1.2, now reads:
> 
> "If timer D fires while the client transaction is in the "Completed"
> state, the client transaction MUST move to the terminated state."
> 
> 
> * We clarified that, when moving to terminated state for client invite
> transactions, you need to delete the transaction immediately.
> 
> * The rules in Section 17.2.3, for matching a request to a
> transaction when the branch ID had the cookie, were not quite
> right. In particular, they were wrong for
> CANCEL. The rules were updated:
> 
>     The
>     request matches a transaction if:
> 
>      1.  the branch parameter in the request
>     is equal to the one in the top Via header field of the 
> request that
>     created the transaction, and
> 
>      2. the sent-by value in the top Via of the
>     request is equal to the one in the request that created the
>     transaction, and
> 
>      3. the method of the request matches the one that created
>     the transaction, except for ACK, where the method of the 
> request that
>     created the transaction is INVITE.
> 
> 
> * Section 18.1.1 led you to believe that if a request was too large
> for UDP, and you resent it, it had to be with TCP. ACtually, it can be
> with any congestion controlled transport (as defined by RFC 2914). We
> also clarified that if you change transports, the Via transport value
> needs to change. (its TCP in any case for our sip stack)
> 

> * Rules for equality matching of From and To header fields were
> absent, even though there were references elsewhere in the text to the
> existence of such rules. They were added. See Sections 20.20 and
> 20.39.
> 
> 
> * The spec was inconsistent on the maximum value of the expires
> Contact parameter and the Expires header field. They were made
> consistent. The maximum value is (2**32)-1.
> 
>
> * Section 20.37, on the Unsupported header field, failed to mention
> that a compact form was defined (the other headers with compact forms
> explicitly mention it in their section 20's). A mention was added.
> 
> * The transaction matching algorithm requires rules for equality of
> Via header fields, but none was defined. A rule was added, see Section
> 20.42.
> 
> * Section 21.4.15 indicated that inclusion of the unsupported
> extensions in a 420 response was a SHOULD. This is inconsistent with
> other text that said it was a MUST. Its a MUST. Section 21.4.15 was
> updated to reflect that.
>
> 
> * The BNF had this construct where there was a list of choices, and
> one of the elements in the choice was optional:
> 
> 
>        foo  =  bar / baz / [boo]
> 
> turns out, this is actually not right, since it allows for an empty
> value for foo, which is not what is intended. Therefore, the text that
> described this convention, and all usages of it in the BNF, were
> removed.
> 
> * The BNF for the user part of the SIP/SIPS URI allowed for an empty
> username and an @ sign; that is "sip:@foo.com" was valid according to
> the BNF. Its not. The BNF for the SIP-URI and SIPS-URI constructions
> was updated to correct this. Here is the change that was made (^
> characters indicate where the change is):
> 
> 
> OLD
>     SIP-URI          =  "sip:" [ userinfo "@" ] hostport
>                         uri-parameters [ headers ]
>     SIPS-URI         =  "sips:" [ userinfo "@" ] hostport
>                         uri-parameters [ headers ]
>     userinfo         =  [ user / telephone-subscriber [ ":" 
> password ]]
>     user             =  *( unreserved / escaped / user-unreserved )
> 
> NEW
>     SIP-URI          =  "sip:" [ userinfo ] hostport
>                                          ^^  delete "@"
>                         uri-parameters [ headers ]
>     SIPS-URI         =  "sips:" [ userinfo ] hostport
>                                           ^^ delete "@"
>                         uri-parameters [ headers ]
>     userinfo         =  ( user / telephone-subscriber ) [ ":" 
> password ] "@"
>                         ^                             ^       
>           ^^^^
>     user             =  1*( unreserved / escaped / user-unreserved )
>                         ^
> 
> * The Accept, Accept-Encoding, and Accept-Language and Allow 
> BNF were all
> updated to allow for the header to have an empty value.
> 
> 
> * The BNF for rfc1123-date was wrong. It was missing a space after the
> comma. It was corrected thusly:
> 
> OLD
> rfc1123-date  =  wkday "," date1 SP time SP "GMT"
> 
> NEW
> rfc1123-date  =  wkday "," SP date1 SP time SP "GMT"
>                             ^^
> 
> * The BNF for the Timestamp header was wrong. It was missing an
> explicit LWS between the the send time and the delay. It was
> added. The BNF was changed thusly:
> 
> 
> OLD
> Timestamp  =  "Timestamp" HCOLON 1*(DIGIT)
>                [ "." *(DIGIT) ] [ delay ]
> 
> NEW
> Timestamp  =  "Timestamp" HCOLON 1*(DIGIT)
>                [ "." *(DIGIT) ] [ LWS delay ]
>                                   ^^^
//lists.cs.columbia.edu/mailman/listinfo/sip-implementors
> 

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From daemon@optimus.ietf.org  Mon Jul 29 11:01:14 2002
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Subject: RE: [Sip] refer-06 question
From: Robert Sparks <rsparks@dynamicsoft.com>
To: patrick.mourot@online.fr
Cc: hisham.khartabil@nokia.com, vijay.kamath@wipro.com, sip@ietf.org
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rfc3265 is authoritative for matters involving sip-events.
Refer only uses it.

You cannot wait for the referenced request to complete. RFC3265
requires the NOTIFY to be sent as soon as the SUBSCRIBE is
accepted (the language Hisham quotes deals with the period between
requesting the subscription and receiving the first notify - that
period needs to be short. Trying to use it to sneak in some processing
time you hope to be short will introduce HERFP troubles later).

An immediate NOTIFY must be sent. If you have the final results
at hand, (through policy for example), that NOTIFY can contain the final
result and be the only NOTIFY sent. Normally, you will have to actually
make the referenced request, and at least two NOTIFYs will be sent.

RjS

On Mon, 2002-07-29 at 04:35, patrick.mourot@online.fr wrote:
> 
> Assuming that you mean RFC3265, there are several contradictory answers.
> I would like to suggest that refer-06 should be clearer on the subject.
> It should describe what a minimal implementation is.
> 
> Patrick
> 
> Quoting hisham.khartabil@nokia.com:
> 
> > I guess it all depends on how you define "immediate" to mean. If its a
> > matter of seconds, then I think its ok to send the one NOTIFY only.
> > 
> > I quote RFC3165
> > 
> > "Until the first NOTIFY message arrives, the subscriber
> >    should consider the state of the subscribed resource to be in a
> >    neutral state.  Documents which define new event packages MUST
> > define
> >    this "neutral state" in such a way that makes sense for their
> >    application (see section 4.4.7.)."
> > 
> > Regards,
> > Hisham
> > 
> > > -----Original Message-----
> > > From: ext Vijay S Kamath [mailto:vijay.kamath@wipro.com]
> > > Sent: Friday, July 26, 2002 3:25 PM
> > > To: patrick.mourot@online.fr
> > > Cc: sip@ietf.org
> > > Subject: RE: [Sip] refer-06 question
> > > 
> > > 
> > > Patrick,
> > > 
> > > You cannot delay the notification until the referenced 
> > > request completes.
> > > Quoting Sec 2.4.4: "The creation of a subscription as defined 
> > > by [1] always results in an
> > > immediate NOTIFY. "
> > > 
> > > So I suggest you follow your second suggestion.
> > > 
> > > Vijay
> > > 
> > > -----Original Message-----
> > > From: patrick.mourot@online.fr [mailto:patrick.mourot@online.fr] 
> > > Sent: Friday, July 26, 2002 5:24 PM
> > > To: Vijay Kamath
> > > Cc: sip@ietf.org
> > > Subject: RE: [Sip] refer-06 question
> > > 
> > > 
> > > vsk,
> > > 
> > > Thanks for the answer.
> > > I understand both cases; but this does not light my way.
> > > Let me put it in other words.
> > > 
> > > can i delay the notification until the referenced request 
> > > completes? leading to send only 1 notification???? (assuming 
> > > timers are OK).
> > > 
> > > Or am i oblige to send a 1rst notification, wait for 
> > > completion of the 
> > > referenced request and send a final notification ?
> > > 
> > > What is mandatory?
> > > 
> > > Patrick
> > > 
> > > 
> > > Quoting Vijay S Kamath <vijay.kamath@wipro.com>:
> > > 
> > > > It can be either 1 or more, but at least 1.
> > > > Qouting Section 2.4.7: "Each NOTIFY must contain a 
> > > Subscription-State 
> > > > header field as defined
> > > >    in [2].  The final NOTIFY sent in response to a REFER 
> > > MUST indicate
> > > >    the subscription has been "terminated" with a reason of 
> > > > "noresource"."
> > > > 
> > > > The final NOTIFY can also be the first NOTIFY.
> > > > 
> > > > Regards... Vijay
> > > > 
> > > > -----Original Message-----
> > > > From: patrick.mourot@online.fr [mailto:patrick.mourot@online.fr]
> > > > Sent: Friday, July 26, 2002 2:48 PM
> > > > To: sip@ietf.org
> > > > Subject: [Sip] refer-06 question
> > > > 
> > > > 
> > > > Sir,
> > > > 
> > > > 
> > > > In 2.4.5 § of refer-06 draft It is stated that a "A minimal, but 
> > > > complete, implementation can respond with a single NOTIFY 
> > > containing 
> > > > either ...the
> > > > body:
> > > > SIP/2.0 200 OK if the reference was successful".
> > > > 
> > > > It seems to be in contradiction with § 3.10 saying that 
> > > > "Alternatively, the subscription might only result in two NOTIFY 
> > > > requests, the immediate NOTIFY and
> > > > the NOTIFY carrying the final result of the reference".
> > > > 
> > > > Do we need 1 or 2 NOTIFY in a minimal implementation?
> > > > 
> > > > Best Regards,
> > > > 
> > > > Patrick
> > > > 
> > > > _______________________________________________
> > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > This list is for NEW development of the core SIP Protocol
> > > > Use sip-implementors@cs.columbia.edu for questions on 
> > > current sip Use 
> > > > sipping@ietf.org for new developments on the application of sip
> > > > 
> > > 
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol
> > > Use sip-implementors@cs.columbia.edu for questions on current sip
> > > Use sipping@ietf.org for new developments on the application of sip
> > > 
> > 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip



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From daemon@optimus.ietf.org  Mon Jul 29 16:40:03 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'BECKMANN MARK'" <mark.beckmann@sal.siemens.de>, <sip@ietf.org>,
        <jdrosen@dynamicsoft.com>
Subject: RE: [Sip] Comments on draft-rosenberg-sip-reg-00
Date: Mon, 29 Jul 2002 15:00:26 -0500
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Mark said:
> Secondly, I think in case an administrator wants a device to 
> re-authenticate, the registration of the corresponding 
> contact should not move to the terminated state. Instead, 
> another event (for example
> "re-authenticate") similar to the "refresh" event should be 
> introduced. When this event occurs, the device is notified 
> and the duration for which the contact remains registered is 
> reduced and if the device fails to re-register, the 
> registration is moved to the terminated state e.g. via the 
> "expired" event.

As I understand it, we want a UA to still work "properly" for a while
after it has been asked to reauthenticate.

If the proxy actually terminates the registration, one might assume it
deletes the associated contact from its storage. An incoming request
(say, INVITE) that arrives at the proxy between this deletion and the
reinsertion of the contact following a new registration from the UA
would then be handled as if the UA was not registered (as in "turned
off").

So, I believe Mark's concerns here are valid and we do need another
state in the FSM.

--
Dean


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From daemon@optimus.ietf.org  Mon Jul 29 16:42:29 2002
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From: "Dean Willis" <dean.willis@softarmor.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>, <sip@ietf.org>,
        <sip-implementors@cs.columbia.edu>
Subject: RE: [Sip] Changes from bis-09 to RFC 3261
Date: Mon, 29 Jul 2002 15:00:26 -0500
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Jonathan wrote:
> There were approximately 150 changes made to bis-09 for the 
> generation of RFC 3261. Coming up with this list was an 
> exercise in extremely detailed and painful review of the 
> document, something which took us a great deal of time. The 
> vast majority of these were typos, comma insertions, 
> indentation fixes, and so on. However, there were a few dozen 
> changes that reflect "bug fixes" which were reported between 
> bis-09 and RFC 3261. Here is a summary of those of any 
> significance whatsoever (you 
> will find most to be trivial):

This seems to be pretty useful. I have posted it to the supplementary
working group web site under the "References" section. See:

http://www.softarmor.com/sipwg/references/changes-bis09-to-3261.html

--
Dean


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From daemon@optimus.ietf.org  Tue Jul 30 00:05:38 2002
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From: "Medhavi Bhatia" <mbhatia@nextone.com>
To: <sip@ietf.org>
Date: Mon, 29 Jul 2002 23:50:44 -0400
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Subject: [Sip] B2B UA definition
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Hi all,

Very many times I have seen a discussion on B2B UA's end merely in a
definition
of a B2BUA:
        "A back-to-back user agent (B2BUA) is a
         logical entity that receives a request and processes it as a
         user agent server (UAS).  In order to determine how the request
         should be answered, it acts as a user agent client (UAC) and
         generates requests.  Unlike a proxy server, it maintains dialog
         state and must participate in all requests sent on the dialogs
         it has established.  Since it is a concatenation of a UAC and
         UAS, no explicit definitions are needed for its behavior."

However I believe this is quite incomplete. For example:

1) A B2BUA behaves similar to a proxy in that it must maintain "timer C" on
outgoing transactions - which is unlike a UAC.
2) It SHOULD maintain max forwards on request in/requests out basis.
Moreover
it SHOULD have a defined relationship between incoming and outgoing
requests.
The way the definition goes, the outgoing request can have completely new
headers,
which eventually will cause a problem in loop detection/ policy and
configuration
at a remote destination. Clearly there may be reasons for new headers, but
certain
headers should be maintained for proper SIP network operation.
3) It has a notion similar to a response context of a proxy, and thus of
forking
requests
 etc etc...

I believe that the SIP rfc describes a "server" role as a stateful/stateless
proxy and
completely specifies it. However a "server" can also be implemented as a
B2BUA,
so it should talk some more about it, or the requirements should be spelled
out
somewhere. A completely loose definition might just lead to deployment
problems
in future.

-Medhavi.



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From daemon@optimus.ietf.org  Tue Jul 30 00:46:55 2002
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Subject: Re: [Sip] B2B UA definition
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Medhavi Bhatia wrote:
> Hi all,
> 
> Very many times I have seen a discussion on B2B UA's end merely in a
> definition
> of a B2BUA:
>         "A back-to-back user agent (B2BUA) is a
>          logical entity that receives a request and processes it as a
>          user agent server (UAS).  In order to determine how the request
>          should be answered, it acts as a user agent client (UAC) and
>          generates requests.  Unlike a proxy server, it maintains dialog
>          state and must participate in all requests sent on the dialogs
>          it has established.  Since it is a concatenation of a UAC and
>          UAS, no explicit definitions are needed for its behavior."
> 
> However I believe this is quite incomplete. For example:
> 
> 1) A B2BUA behaves similar to a proxy in that it must maintain "timer C" on
> outgoing transactions - which is unlike a UAC.
> 2) It SHOULD maintain max forwards on request in/requests out basis.
> Moreover
> it SHOULD have a defined relationship between incoming and outgoing
> requests.
> The way the definition goes, the outgoing request can have completely new
> headers,
> which eventually will cause a problem in loop detection/ policy and
> configuration
> at a remote destination. Clearly there may be reasons for new headers, but
> certain
> headers should be maintained for proper SIP network operation.
> 3) It has a notion similar to a response context of a proxy, and thus of
> forking
> requests
>  etc etc...

I think these things greatly overconstrain a b2bua. If we follow this 
path to the logical conclusion, a b2bua is just another way to implement 
a proxy. For example, is it not permissible to create a b2bua that, when 
it receives one request, generates several? Or when it receives a FOO 
request, emits a BAR request?

A b2bUA must act as a proper UAS on one side, and a proper UAC on the 
other. That should be enough. You describe some particular 
application(s) of a b2bua, where the inbound and outbound dialogs are 
linked in a way that is at least somewhat transparent to other devices. 
I think there are many b2bua applications where that is not the case at 
all. For example, can I not create a b2bua, where the downstream devices 
cannot tell that the upstream devices exist?


> 
> I believe that the SIP rfc describes a "server" role as a stateful/stateless
> proxy and
> completely specifies it. However a "server" can also be implemented as a
> B2BUA,
> so it should talk some more about it, or the requirements should be spelled
> out
> somewhere. A completely loose definition might just lead to deployment
> problems
> in future.
> 
> -Medhavi.
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 




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From daemon@optimus.ietf.org  Tue Jul 30 00:51:34 2002
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To: "Medhavi Bhatia" <mbhatia@nextone.com>
Cc: <sip@ietf.org>
Subject: [Sip] B2B UA definition
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Is there any reason at all to say what a B2BUA is
and/or needs to do? *Anything* can be a B2BUA, and
I don't see why its behavior should be defined any
more than defining the nature of my procmail
recipes wrt SMTP.

It was a good thing to illustrate that a B2BUA
could exist, but I just don't see that we should
go overboard...

		Mike

Medhavi Bhatia writes:
 > Hi all,
 > 
 > Very many times I have seen a discussion on B2B UA's end merely in a
 > definition
 > of a B2BUA:
 >         "A back-to-back user agent (B2BUA) is a
 >          logical entity that receives a request and processes it as a
 >          user agent server (UAS).  In order to determine how the request
 >          should be answered, it acts as a user agent client (UAC) and
 >          generates requests.  Unlike a proxy server, it maintains dialog
 >          state and must participate in all requests sent on the dialogs
 >          it has established.  Since it is a concatenation of a UAC and
 >          UAS, no explicit definitions are needed for its behavior."
 > 
 > However I believe this is quite incomplete. For example:
 > 
 > 1) A B2BUA behaves similar to a proxy in that it must maintain "timer C" on
 > outgoing transactions - which is unlike a UAC.
 > 2) It SHOULD maintain max forwards on request in/requests out basis.
 > Moreover
 > it SHOULD have a defined relationship between incoming and outgoing
 > requests.
 > The way the definition goes, the outgoing request can have completely new
 > headers,
 > which eventually will cause a problem in loop detection/ policy and
 > configuration
 > at a remote destination. Clearly there may be reasons for new headers, but
 > certain
 > headers should be maintained for proper SIP network operation.
 > 3) It has a notion similar to a response context of a proxy, and thus of
 > forking
 > requests
 >  etc etc...
 > 
 > I believe that the SIP rfc describes a "server" role as a stateful/stateless
 > proxy and
 > completely specifies it. However a "server" can also be implemented as a
 > B2BUA,
 > so it should talk some more about it, or the requirements should be spelled
 > out
 > somewhere. A completely loose definition might just lead to deployment
 > problems
 > in future.
 > 
 > -Medhavi.
 > 
 > 
 > 
 > _______________________________________________
 > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
 > This list is for NEW development of the core SIP Protocol
 > Use sip-implementors@cs.columbia.edu for questions on current sip
 > Use sipping@ietf.org for new developments on the application of sip

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Jul 30 02:47:37 2002
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From: "Cullen Jennings" <fluffy@cisco.com>
To: "Medhavi Bhatia" <mbhatia@nextone.com>, <sip@ietf.org>
Subject: RE: [Sip] B2B UA definition
Date: Mon, 29 Jul 2002 23:25:41 -0700
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I think that a B2BUA should meet the requirements of a UAC on one side and a
UAS on the other. I am fine with it resetting max forwards and all the other
stuff you mention. If you consider a call from a UA to gateway that go
across the PSTN to another gateway to a UA, you realize the PSTN is one
large B2BUA. I don't think B2BUAs would benefit much from further definition
than it has today.

However, you could get me going about what a "transparent" B2BUA is :-)

Cullen

> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org]On Behalf Of Medhavi
> Bhatia
> Sent: Monday, July 29, 2002 8:51 PM
> To: sip@ietf.org
> Subject: [Sip] B2B UA definition
>
>
> Hi all,
>
> Very many times I have seen a discussion on B2B UA's end merely in a
> definition
> of a B2BUA:
>         "A back-to-back user agent (B2BUA) is a
>          logical entity that receives a request and processes it as a
>          user agent server (UAS).  In order to determine how the request
>          should be answered, it acts as a user agent client (UAC) and
>          generates requests.  Unlike a proxy server, it maintains dialog
>          state and must participate in all requests sent on the dialogs
>          it has established.  Since it is a concatenation of a UAC and
>          UAS, no explicit definitions are needed for its behavior."
>
> However I believe this is quite incomplete. For example:
>
> 1) A B2BUA behaves similar to a proxy in that it must maintain
> "timer C" on
> outgoing transactions - which is unlike a UAC.
> 2) It SHOULD maintain max forwards on request in/requests out basis.
> Moreover
> it SHOULD have a defined relationship between incoming and outgoing
> requests.
> The way the definition goes, the outgoing request can have completely new
> headers,
> which eventually will cause a problem in loop detection/ policy and
> configuration
> at a remote destination. Clearly there may be reasons for new headers, but
> certain
> headers should be maintained for proper SIP network operation.
> 3) It has a notion similar to a response context of a proxy, and thus of
> forking
> requests
>  etc etc...
>
> I believe that the SIP rfc describes a "server" role as a
> stateful/stateless
> proxy and
> completely specifies it. However a "server" can also be implemented as a
> B2BUA,
> so it should talk some more about it, or the requirements should
> be spelled
> out
> somewhere. A completely loose definition might just lead to deployment
> problems
> in future.
>
> -Medhavi.
>
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Tue Jul 30 03:37:57 2002
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From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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Subject: Re: [Sip] draft-ietf-sip-sctp-03.txt: comment on unordered delivery
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Hi,

By reading the paragraph you refer to, it is pretty clear that the
unordered flag should not be set in the case you are talking about.

Thanks,

Gonzalo

hbhondwe@hss.hns.com wrote:
> 
> Hi,
> 
> In section 4.1 Mapping of SIP Transactions into Streams,
> it says
> "..a SIP entity SHOULD send every SIP message
>    (request or response) over stream zero with the unordered flag set..."
> 
> Are there no SIP extensions which may require ordered delivery  from
> SCTP(sip-isup mapping for instance)?
> or is the 'SHOULD'  supposed to accommodate such exceptions?
> 
> Later in the same section, it is said "..Some applications introduce an
> extra layer between the transport
>    layer and SIP (e.g., compression and/or encryption). This extra layer
>    sometimes requires ordered delivery of messages from the transport
>    layer (e.g., TLS [6])."
> 
> The section then recommends using same stream for maintaining
> sequence...but no mention is made about whether the unordered flag should
> be set or not. To maintain sequence, in addition to choosing the same
> stream, will it not be required that messages be also sent in sequence
> within a stream(by not setting the unordered flag)?
> 
> regards
> Harsh Bhondwe
> Hughes Software Systems
> 
> This message is proprietary to Hughes Software Systems Limited (HSS) and is
> intended solely for the use of the individual to whom it is addressed.  It
> may contain privileged or confidential information and should not be
> circulated or used for any purpose other than for what it is intended.  If
> you have received this message in error, please notify the originator
> immediately.  If you are not the intended recipient, you are notified that
> you are strictly prohibited from using, copying, altering, or disclosing
> the contents of this message.  HSS accepts no responsibility for loss or
> damage arising from the use of the information transmitted by this email
> including damage from virus.
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo

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From daemon@optimus.ietf.org  Tue Jul 30 10:56:08 2002
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Date: Tue, 30 Jul 2002 10:26:58 -0400
From: "David R. Oran" <oran@cisco.com>
To: Cullen Jennings <fluffy@cisco.com>, Medhavi Bhatia <mbhatia@nextone.com>,
        sip@ietf.org
Subject: RE: [Sip] B2B UA definition
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--On Monday, July 29, 2002 11:25 PM -0700 Cullen Jennings 
<fluffy@cisco.com> wrote:

>
> I think that a B2BUA should meet the requirements of a UAC on one side
> and a UAS on the other. I am fine with it resetting max forwards and all
> the other stuff you mention. If you consider a call from a UA to gateway
> that go across the PSTN to another gateway to a UA, you realize the PSTN
> is one large B2BUA. I don't think B2BUAs would benefit much from further
> definition than it has today.
>
> However, you could get me going about what a "transparent" B2BUA is :-)
>
That's easy. A transparent B2BUA is one whose index of refraction is close 
to zero.

	:-)

Transparent is one of the most misused terms in our entire industry, right 
up there with "compatible", and "seamless".

Banning this terminology would do everyone a big service. Oops, then 
there's most misused term in the telecom industry..."service".

Avoiding these terms would go a long way to keeping me from getting 
confused.
	:-) :-)

Dave.

> Cullen


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This list is for NEW development of the core SIP Protocol
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From daemon@optimus.ietf.org  Wed Jul 31 04:21:12 2002
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Date: Wed, 31 Jul 2002 09:45:15 +0200 (CEST)
From: Salvador Rey Calatayud <salreyca@teleco.upv.es>
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Hi everyone,

I resubmit this question since it has got no answer yet,

as for the transaction state diagrams, and section 14.1 of RFC 3261, a UAS
that had received an INVITE(not blank) cannot send another one, and thus start
a new offer/answer cycle, until it sends a 200ok, or receives the ACK for a
300-699 response. why is the rule different for 200ok than for other responses?

section 14.1 RFC 3261
"2. If there is an ongoing INVITE server transaction, the TU MUST wait
until the transaction reaches the confirmed or terminated state before initiating
the new INVITE."

I think UPDATE and this are not alligned since the UPDATE specification
says that another offer cannot be sent until a previous 1xxrel containing a
descriptor is acknowledged with a PRACK (playing the role of ACK to 200ok
in basic SIP).

I include section 5.1 of UPDATE-02
"and for the callee:
o If the UPDATE is being sent before the completion of the
INVITE transaction, and the initial INVITE contained an offer,
the UPDATE cannot be sent unless the callee has generated an
answer in a reliable provisional response, !!has received a
PRACK for that reliable provisional response!!, has not received
any requests (PRACK or UPDATE) with offers that it has not
answered, and has not sent any UPDATE requests containing
offers that have not been answered."

comments?

regards,
Salva

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Hi
"Note that a UAC MUST NOT initiate a new INVITE transaction within a
   dialog while another INVITE transaction is in progress in either
   direction."
This is only about new INVITE transactions within an "existing" dialog.
If the UAS has sent a 200 ok, a new INVITE transaction within a dialog is
just a
(re) INVITE.

So your other question was, why the rule for other responses than 200ok are
different. Are they? It is just, that the TU is responsible for the ACK
handling and not the transaction layer itself. And that is, because positive
final response are end to end acknowleged an negative ones hop by hop.




----- Original Message -----
From: "Salvador Rey Calatayud" <salreyca@teleco.upv.es>
To: <sip@ietf.org>
Sent: Wednesday, July 31, 2002 9:45 AM
Subject: [Sip] offer/answer exchange


> Hi everyone,
>
> I resubmit this question since it has got no answer yet,
>
> as for the transaction state diagrams, and section 14.1 of RFC 3261, a UAS
> that had received an INVITE(not blank) cannot send another one, and thus
start
> a new offer/answer cycle, until it sends a 200ok, or receives the ACK for
a
> 300-699 response. why is the rule different for 200ok than for other
responses?
>
> section 14.1 RFC 3261
> "2. If there is an ongoing INVITE server transaction, the TU MUST wait
> until the transaction reaches the confirmed or terminated state before
initiating
> the new INVITE."
>
> I think UPDATE and this are not alligned since the UPDATE specification
> says that another offer cannot be sent until a previous 1xxrel containing
a
> descriptor is acknowledged with a PRACK (playing the role of ACK to 200ok
> in basic SIP).
>
> I include section 5.1 of UPDATE-02
> "and for the callee:
> o If the UPDATE is being sent before the completion of the
> INVITE transaction, and the initial INVITE contained an offer,
> the UPDATE cannot be sent unless the callee has generated an
> answer in a reliable provisional response, !!has received a
> PRACK for that reliable provisional response!!, has not received
> any requests (PRACK or UPDATE) with offers that it has not
> answered, and has not sent any UPDATE requests containing
> offers that have not been answered."
>
> comments?
>
> regards,
> Salva
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>


_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From daemon@optimus.ietf.org  Wed Jul 31 08:14:21 2002
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Thread-Topic: draft-ietf-sip-privacy-general-04
Thread-Index: AcI4X37UsMlUXCg7QeuArWxcskpDgAAADnAQ
From: "ShavinderPal Singh" <ShavinderPal.Singh@ssdsi.com>
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> Hi,
> I am not able to find draft-ietf-sip-privacy-general-04 on ietf.org 
> 
> Can anyone help me find it on the net somewhere. I searched on the site. I could find only draft-ietf-sip-privacy-general-01.
> 
> Thanks & Rgds
> s
> 
> Samsung SDS Infotech. P Ltd.
> Ph.: 091-124-6383474 (e 254)
> _________________________________
> "I am not suffering from insanity, 
> I am enjoying every single minute of it!".
> 


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This list is for NEW development of the core SIP Protocol
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From daemon@optimus.ietf.org  Wed Jul 31 08:15:44 2002
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Thread-Topic: Method name in Request Line and CSeq.
Thread-Index: AcI4f+vRcXv30VGBRf+FvVJil3tHKg==
From: "ShavinderPal Singh" <ShavinderPal.Singh@ssdsi.com>
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Subject: [Sip] Method name in Request Line and CSeq.
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Hi,

I have a very trivial query. Why do we need to keep the METHOD name both in the Request Line and the CSeq. Header, when always it will be same, in case of a REQUEST.
In case of a response - there is nothing as called as a 'request line', so in any ways we have to look into the CSeq Header to find out the METHOD make out the fact that for which request was this response sent.

Please clarify my doubts.

Thanks

Rgds
s

Samsung SDS Infotech. P Ltd.
Ph.: 091-124-6383474 (e 254)
_________________________________
"I am not suffering from insanity, 
I am enjoying every single minute of it!".



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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From daemon@optimus.ietf.org  Wed Jul 31 08:19:01 2002
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Subject: [Sip] I-D ACTION:draft-khartabil-sip-congestionsafe-ci-00.txt
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: Congestion safety and Content Indirection
	Author(s)	: H. Khartabil
	Filename	: draft-khartabil-sip-congestionsafe-ci-00.txt
	Pages		: 15
	Date		: 25-Jul-02
	
The Session Initiation Protocol allows the use of UDP for transport 
of SIP messages. Baseline SIP does not allow congestion control for 
messages larger than MTU of a certain link nor does it allow end 
points to specify their maximum acceptable message size, regardless 
of the underlying transport protocol. 
This document combines two, namely congestion safely and Content-
Indirection Mechanism into the one document and presents scenarios 
where the 2 could be combined. It also introduces extensions to SIP 
that allows end points to specify their maximum acceptable message 
size.

A URL for this Internet-Draft is:
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From daemon@optimus.ietf.org  Wed Jul 31 09:39:36 2002
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Hi,

I have a comment on the privacy services and their support on proxy or
other servers.

If privacy services defined in terms of different levels can be
supported/not supported by different proxy servers, then one can also
assume that probably these may be activated/deactivated on existing servers
in the network. If we accept this scenario, then the draft needs to address
situations like the following:
1.A proxy server which is performing levels A out of a total levels A,B
requested by the UA, now is ready to activate level B. What should it do to
the session for which it is only performing level A?(case addition of a
privacy level)
2.A proxy server which is performing levels A out of a total levels A,B
requested by the UA, now wants  to deactive level B. What should it do to
the session for which it is only performing level A?(case addition of a
privacy level) Should it reroute future messages through another server
which supports both A,B. Or should it only stop removing the prev-value so
that subsequent servers start to perform the privacy service A. Or should
deactivation of a level on a server be allowed only when the level in
question is not in use.

IMHO such scenarios may be valid ones, and some guidelines or explanations
should be provided in the draft to cover them.

Comments?

regards
Harsh Bhondwe
Hughes Software Systems






This message is proprietary to Hughes Software Systems Limited (HSS) and is
intended solely for the use of the individual to whom it is addressed.  It
may contain privileged or confidential information and should not be
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From daemon@optimus.ietf.org  Wed Jul 31 09:58:21 2002
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Hi,

I have a comment on the privacy services and their support on proxy or
other servers.

If privacy services defined in terms of different levels can be
supported/not supported by different proxy servers, then one can also
assume that probably these may be activated/deactivated on existing servers
in the network. If we accept this scenario, then the draft needs to address
situations like the following:
1.A proxy server which is performing levels A out of a total levels A,B
requested by the UA, now is ready to activate level B. What should it do to
the session for which it is only performing level A?(case activation of a
privacy level)
2.A proxy server which is performing levels A out of a total levels A,B
requested by the UA, now wants  to deactive level A. What should it do to
the session for which it is only performing level A?(case deactivation of a
privacy level) Should it reroute future messages through another server
which supports both A,B. Or should it only stop removing the prev-value so
that subsequent servers start to perform the privacy service A. Or should
deactivation of a level on a server be allowed only when the level in
question is not in use.

IMHO such scenarios may be valid ones, and some guidelines or explanations
should be provided in the draft to cover them.

Comments?

regards
Harsh Bhondwe
Hughes Software Systems






This message is proprietary to Hughes Software Systems Limited (HSS) and is
intended solely for the use of the individual to whom it is addressed.  It
may contain privileged or confidential information and should not be
circulated or used for any purpose other than for what it is intended.  If
you have received this message in error, please notify the originator
immediately.  If you are not the intended recipient, you are notified that
you are strictly prohibited from using, copying, altering, or disclosing
the contents of this message.  HSS accepts no responsibility for loss or
damage arising from the use of the information transmitted by this email
including damage from virus.



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Wed Jul 31 12:18:55 2002
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Subject: [Sip] SIP Congestion Control
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Hi,

Does SIP support congestion/flow control methods during application level
processing overload? (e.g. something equivalent to ISUP's ACC)

This is typically for scenarios where there could be a burst of calls (possibly
due to some media stimulated event) and hence a burst of SIP messages. The
target of these messages could either be a proxy server, voicemail server or
some other media server.

I am assuming there are sufficient resources in the IP network and no packets
will be dropped.

From going through RFC3261 SIP seems to rely on the congestion controls of the
transport protocols.
draft-willis-sip-congestsafe-00.txt also seems to address the case where a
transport network protocols like TCP and SCTP provide feedback control.

Thanks,
Amish





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From daemon@optimus.ietf.org  Wed Jul 31 13:06:32 2002
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From: "Naidu, Venkata" <Venkata.Naidu@Marconi.com>
To: "'amish.chana@orange.co.uk'" <amish.chana@orange.co.uk>, sip@ietf.org
Subject: RE: [Sip] SIP Congestion Control
Date: Wed, 31 Jul 2002 12:27:49 -0400
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Amish,

-> I am assuming there are sufficient resources in the IP 
-> network and no packets will be dropped.

  Then where is the congestion in the network?
 
-> From going through RFC3261 SIP seems to rely on the 
-> congestion controls of the
-> transport protocols.
-> draft-willis-sip-congestsafe-00.txt also seems to address 
-> the case where a
-> transport network protocols like TCP and SCTP provide 
-> feedback control.

  I don't think congestion control, flow control methods
  are required in session/application level protocols
  when they depend on well tested, extensively reserached
  transport protocols (like TCP and respective extensions).

  Looks like every higher layer protocols are trying to
  solve their own congetion/flow control. Yes! I agree
  that there must be some corner cases where you can
  optimize/improve protocols (like SIP, BGP etc) when there 
  is much stress. But the problems you are trying to solve
  are very corner cases when you implement congestion/flow
  control in SIP, BGP etc

  I feel that this is duplicate work unless SIP is going
  achieve good preformance improvement with it's own
  congestion/flow control mechanisms when compared to
  best TCP extension(s).

--
Venkata.

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From daemon@optimus.ietf.org  Wed Jul 31 13:59:44 2002
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Subject: Re: [Sip] SIP Congestion Control
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 Wouldn't this sort of load shedding or load balancing issue be best dealt
with at the application and not the protocol layer.






amish.chana@orange.co.uk@ietf.org on 07/31/2002 11:34:51 AM

Sent by:  sip-admin@ietf.org


To:   sip@ietf.org
cc:
Subject:  [Sip] SIP Congestion Control



Hi,

Does SIP support congestion/flow control methods during application level
processing overload? (e.g. something equivalent to ISUP's ACC)

This is typically for scenarios where there could be a burst of calls
(possibly
due to some media stimulated event) and hence a burst of SIP messages. The
target of these messages could either be a proxy server, voicemail server
or
some other media server.

I am assuming there are sufficient resources in the IP network and no
packets
will be dropped.

From going through RFC3261 SIP seems to rely on the congestion controls of
the
transport protocols.
draft-willis-sip-congestsafe-00.txt also seems to address the case where a
transport network protocols like TCP and SCTP provide feedback control.

Thanks,
Amish





*******************************************************************************

Important.
Confidentiality: This communication is intended for the above-named person
and
may be confidential and/or legally privileged. Any opinions expressed in
this
communication are not necessarily those of the company. If it has come to
you
in error you must take no action based on it, nor must you copy or show it
to
anyone; please delete/destroy and inform the sender immediately.

Monitoring/Viruses
Orange may monitor all incoming and outgoing emails in line with current
legislation.  Although we have taken steps to ensure that this email and
attachments are free from any virus, we advise that in keeping with good
computing practice the recipient should ensure they are actually virus
free.

Orange PCS Limited is a subsidiary of Orange SA and is registered in
England No
2178917, with its address at St James Court, Great Park Road, Almondsbury
Park,
Bradley Stoke, Bristol BS32 4QJ.
*******************************************************************************



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip




_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip



From daemon@optimus.ietf.org  Wed Jul 31 20:35:28 2002
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Reply-To: <JackLix@attbi.com>
From: "Jack W. Lix" <JackLix@attbi.com>
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Date: Wed, 31 Jul 2002 17:27:05 -0700
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Subject: [Sip] caller preferences and RFC 2533
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Hi,

I've been reviewing
http://search.ietf.org/internet-drafts/draft-ietf-sip-message-06.txt and
have a couple questions.

First, in section 5.2 the following paragraph is found:

> The proxy then takes each URI in the target set (the set of URI it is
> going to proxy to), and obtains there capabilities as an RFC 2533
> formatted feature set. If target URI was obtained through a
> registration, the proxy computes the feature set by taking all
> Contact URI parameters except for the q and expires parameters, and
> converting them to RFC 2533 syntax using the rules of Section 6.1.

Should the word "there" in the 1st sentence be their or it's?  Or, should
the sentence read "and there obtains capabilities"?

Second, the text makes many references to RFC 2533 and that is fine for
describing the intended behaviour.  However, it implies that you MUST
translate the headers into RFC 2533 format.  Doesn't that impose an
implementation rather than define the protocol?  Is there any reason that I
really have to use RFC 2533?

Thanks,

Jack W. Lix
Wind River Systems



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