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Subject: RE: [Sip] Jon Peterson exiting Chair role in SIP WG
Date: Tue, 1 Jul 2003 08:49:53 -0400
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Thread-Topic: [Sip] Jon Peterson exiting Chair role in SIP WG
Thread-Index: AcM/VqMtDblbv3cuSVK+4mYCTzKCeQAd8pww
From: "Roy, Radhika R, ALABS" <rrroy@att.com>
To: "Dean Willis" <dean.willis@softarmor.com>, <sip@ietf.org>
Cc: <rohan@cisco.com>, <jon.peterson@neustar.biz>, <mankin@psg.com>
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Inline [RRR]

I believe
Allison Mankin will be continuing as the primary AD for SIP, but we can
expect that Jon will be very able to assist in the IESG review and
supervision of our work.

[RRR] We also believe that it would be enormously helpful, if Jon
continues to provides his insights in furthering the technical
discussions as one of us as he has been doing for a long time, if
possible.

Radhika R. Roy
rrroy@att.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



From exim@www1.ietf.org  Tue Jul  1 10:06:37 2003
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From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
To: "'Hearty, John'" <John.Hearty@Level3.com>, sip@ietf.org,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Subject: RE: [Sip] Stuck in Proceeding after Cancel
Date: Mon, 30 Jun 2003 07:55:04 +0200
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Hi,

Isn't the timer you are talking about timer B? I agree that it would be more clear to write "timer B" instead of "64*T1 seconds".

I also think it is important to remember that timer B may expire AFTER CANCEL has been sent, but BEFORE 487 has been received, and since the transaction will be terminated there is no guarantee an ACK will be sent when 487 eventually arrives. But, I guess this is similar to any case where the final response is sent too late by the UAS...?

Regards,

Christer Holmberg
Ericsson Finland



>  I found the following text that covers this scenario, so it 
> may not be
> considered a bug.  I don't particularly like the idea of an unnamed
> timer though, and the below text could be a bit more explicit 
> about when
> to start the unnamed timer (upon sending Cancel).  I would appreciate
> this scenario being added to the existing bugzilla bug so it is also
> included in some future clarification of the Invite client transaction
> section.
> 
>    Note that both the transaction corresponding to the 
> original request
>    and the CANCEL transaction will complete independently.  However, a
>    UAC canceling a request cannot rely on receiving a 487 (Request
>    Terminated) response for the original request, as an RFC 2543-
>    compliant UAS will not generate such a response.  If there is no
>    final response for the original request in 64*T1 seconds (T1 is
> 
> 
> 
> 
> Rosenberg, et. al.          Standards Track                   
>  [Page 54]
>  
> 
> RFC 3261            SIP: Session Initiation Protocol          
>  June 2002
> 
> 
>    defined in Section 17.1.1.1), the client SHOULD then consider the
>    original transaction cancelled and SHOULD destroy the client
>    transaction handling the original request.
> 
> 
> John Hearty
> Level3
> 
> > -----Original Message-----
> > From: Hearty, John
> > Sent: Thursday, June 26, 2003 12:47 PM
> > To: sip@ietf.org
> > Subject: [Sip] Stuck in Proceeding after Cancel
> > 
> > We have come across what may be a bug in RFC 3261.  The 
> following bug
> is
> > closely related, but it is not clear to me it is the same.
> > 
> > http://bugs.sipit.net/sipwg/show_bug.cgi?id=706
> > 
> > The issue involves the Invite client transaction state machine.  As
> the
> > above bug describes, there is no timer to exit the Proceeding state.
> > This is normally fine, as it is an application decision 
> when give up.
> > 
> > However, when the application does give up and sends a Cancel, we
> still
> > rely on the UAS to return the forced 487 to properly exit the
> Proceeding
> > state.  If the 487 (or any final response) is never returned for
> > whatever reason, we remain stuck in Proceeding.
> > 
> > I am not clear if there should be a timer specified in the spec to
> allow
> > exit from this state in this particular scenario, or if this should
> also
> > be considered an application layer problem and covered under the
> > existing bug.
> > 
> > John Hearty
> > Level3
> > 
> > _______________________________________________
> > 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 exim@www1.ietf.org  Tue Jul  1 10:54:29 2003
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From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
To: "'Hearty, John'" <John.Hearty@Level3.com>, sip@ietf.org,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Subject: RE: [Sip] Stuck in Proceeding after Cancel
Date: Tue, 1 Jul 2003 07:20:54 +0200 
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Hi John,

>From the duration, it sounds like timer B, though I didn't want to
>presume anything.  However, the text is not clear when said timer gets
>set.  Also, if it actually is timer B which is set when the original
>request is sent, it could be stopped or ignored (exact action 
>regarding the timer is not specified) once a provisional response is received.

Correct. However, I can't find any text saying that, and the bug description in bugzilla says it's pretty much an implementation issue. 

However, is it even said somewhere that/if/how timer B will be stopped when we receive a FINAL response, and go to the completed state? At that point we DO start timer D, but if we haven't stopped timer B it will most likely expire before timer D, so there would be no need for timer D in the first place, would there?

>The below text appears to suggest said timer continues to run in this
>specific scenario.
> 
>This just seems like another of the countless clarifications that the
>evolving specification may need.  If I had to make the 
>clarification, it would be to reset timer B for the Invite Client Transaction 
>upon sending a Cancel for that Invite.

I agree. Reset it, or not touch it at all, but certainly not stop it.

>That would fix it, and provide an equal time for a 487 response that a provisional response or other non-2xx final
>response had.  I think 487 is a special case compared to other non-2xx final responses.

Also, I wonder if it would be a good idea to also reset timer B when we get a provisional response, if we use PRACK, to equal time for the INVITE final response and PRACK final response.

Regards,

Christer Holmberg
Ericsson Finland




> 
> John Hearty
> Level3
> 
> > -----Original Message-----
> > From: Christer Holmberg (JO/LMF)
> [mailto:christer.holmberg@ericsson.com]
> > Sent: Sunday, June 29, 2003 11:55 PM
> > To: Hearty, John; sip@ietf.org; Jonathan Rosenberg
> > Subject: RE: [Sip] Stuck in Proceeding after Cancel
> > 
> > 
> > Hi,
> > 
> > Isn't the timer you are talking about timer B? I agree that it would
> be
> > more clear to write "timer B" instead of "64*T1 seconds".
> > 
> > I also think it is important to remember that timer B may 
> expire AFTER
> > CANCEL has been sent, but BEFORE 487 has been received, and 
> since the
> > transaction will be terminated there is no guarantee an ACK will be
> sent
> > when 487 eventually arrives. But, I guess this is similar 
> to any case
> > where the final response is sent too late by the UAS...?
> > 
> > Regards,
> > 
> > Christer Holmberg
> > Ericsson Finland
> > 
> > 
> > 
> > >  I found the following text that covers this scenario, so it
> > > may not be
> > > considered a bug.  I don't particularly like the idea of 
> an unnamed
> > > timer though, and the below text could be a bit more explicit
> > > about when
> > > to start the unnamed timer (upon sending Cancel).  I would
> appreciate
> > > this scenario being added to the existing bugzilla bug so 
> it is also
> > > included in some future clarification of the Invite client
> transaction
> > > section.
> > >
> > >    Note that both the transaction corresponding to the
> > > original request
> > >    and the CANCEL transaction will complete 
> independently.  However,
> a
> > >    UAC canceling a request cannot rely on receiving a 487 (Request
> > >    Terminated) response for the original request, as an RFC 2543-
> > >    compliant UAS will not generate such a response.  If 
> there is no
> > >    final response for the original request in 64*T1 seconds (T1 is
> > >
> > >
> > >
> > >
> > > Rosenberg, et. al.          Standards Track
> > >  [Page 54]
> > >
> > >
> > > RFC 3261            SIP: Session Initiation Protocol
> > >  June 2002
> > >
> > >
> > >    defined in Section 17.1.1.1), the client SHOULD then 
> consider the
> > >    original transaction cancelled and SHOULD destroy the client
> > >    transaction handling the original request.
> > >
> > >
> > > John Hearty
> > > Level3
> > >
> > > > -----Original Message-----
> > > > From: Hearty, John
> > > > Sent: Thursday, June 26, 2003 12:47 PM
> > > > To: sip@ietf.org
> > > > Subject: [Sip] Stuck in Proceeding after Cancel
> > > >
> > > > We have come across what may be a bug in RFC 3261.  The
> > > following bug
> > > is
> > > > closely related, but it is not clear to me it is the same.
> > > >
> > > > http://bugs.sipit.net/sipwg/show_bug.cgi?id=706
> > > >
> > > > The issue involves the Invite client transaction state machine.
> As
> > > the
> > > > above bug describes, there is no timer to exit the Proceeding
> state.
> > > > This is normally fine, as it is an application decision
> > > when give up.
> > > >
> > > > However, when the application does give up and sends a 
> Cancel, we
> > > still
> > > > rely on the UAS to return the forced 487 to properly exit the
> > > Proceeding
> > > > state.  If the 487 (or any final response) is never returned for
> > > > whatever reason, we remain stuck in Proceeding.
> > > >
> > > > I am not clear if there should be a timer specified in 
> the spec to
> > > allow
> > > > exit from this state in this particular scenario, or if this
> should
> > > also
> > > > be considered an application layer problem and covered under the
> > > > existing bug.
> > > >
> > > > John Hearty
> > > > Level3
> > > >
> > > > _______________________________________________
> > > > 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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Subject: [Sip] Simple question about NOTIFY
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Hi All,
 
As I am reading through the two RFCs 3261, and 3265, I am just wondering if a SIP NOTIFY can be forked? My guess is it cannot be forked, but I would like to double check :-). Also if it can be forked, can someone please explain how will that work, since NOTIFY would be sent in a dialog, and according to the rules set in section 12 of RFC 3261, the Request-URI would need to be set to the remote target (which is the contact address of the subscriber), and a Proxy is just supposed to forward to that target.
 
Appreciate your help.
 
Thanks and Regards
- Ajay

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Subject: [Sip] Re: Callerprefs: Additional Nit/editorial comments
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Mary,

Now that I've finished splitting the document, I've been able to
incorporate these nits which are still applicable. Responses below.

Mary Barnes wrote:
> I also did a detailed review of draft-ietf-sip-callerprefs-08.txt and beyond
> some of the points already made by Ben and Bob, I had the following
> additional minor NITs/Editorial comments.  There also a couple of  points
> that I'll put in separate threads as they could invoke some additional
> discussion.  Also, I did try to avoid duplicating comments made by Ben and
> Bob and apologize if I have inadvertently duplicated anything.
> 
> Footing: 
> - there is no "." in the "et" of the phrase "et al."

This is an error in the tool I was using. I've converted to xml2rfc in
the process of the split, and so this problem is now gone.

> 
> Section 3, page 7: 
> - Feature Preferences: "described" -> "describe" 

Fixed.

> - Target Set: "URI" -> "URI(s)"

Fixed.

> 
> Section 5: 
> - Para 5: "instant of time" -> "instance of time"

Fixed.

> 
> Section 7.3:
> - Last para: "featuer" -> "feature"

Fixed.

> 
> Section 7.4:
> - Para 5: I know you have an example further down in section 7.4.1, but I'm
> thinking an example after this paragraph might be useful.  I had to read the
> last few sentences many several times and I'm still not absolutely certain I
> understand it given that the example in 7.4.1 has the "require" flag set.
> So, I think another example here (or forward referenced to one in 7.4.1)
> would be extremely helpful.  

I actually added a lot of text in the UA Behavior section of the new 
draft to elaborate the behaviors as they relate to "require" and 
"explicit". I think that this text, in conjunction with the example, 
should suffice. The use cases spec also provides a lot more examples, 
so I don't want to put too many into caller prefs itself.

> 
> Section 9.6 Automata: 
> - Would IVR be considered here as well?  

Yes. I added a mention of it.

> 
> Section 9.12 Priority:
> - I would suggest changing the definition of "emergency" to "The device
> supports calls in the case of an emergency situation" as I think the phrase
> "emergency calls" tends to so closely associated with an E911 type of call,
> whereas I think this is meant to be in terms of the device being one that
> user wants to be the primary contact for personal emergency situations.

Good point. Fixed.

> 
> Section 9.15 Schemes:
> - Example of typical use: "Choosing get" -> "Choosing to get"

Fixed.

> 
> Section 9.17 Message Server:
> - Is this category intended to also apply to IVR? Or, would IVR just be
> considered automata?   

No, an IVR is not a message server.

> 
> Section 9.18 Is Focus:
> - this is just a consistency thing, but it seems that this should just be
> "Focus" rather than "isfocus" or 9.17 should be "ismsgserver". 

Its a legacy thing. We've been using isfocus in the conferencing stuff
for a while, and people are already using that specific name.

> 
> Section 17 Informative References:
> - [27] needs to be updated with the RFC number: RFC 3515

This reference isn't there any more. There is no longer a need to
enumerate the packages. I merely reference the IANA registry.

Thanks,
Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Scientist                             Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
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From exim@www1.ietf.org  Wed Jul  2 03:43:16 2003
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Date: Wed, 02 Jul 2003 03:40:29 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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I dont think I ever answered this.

George Foti (LMC) wrote:

> Quoting the 08 draft  section 7.4 (Matching):
> 
> Next, the proxy applies the predicates associated with the 
> Accept-Contact header field. For each contact that remains in the target 
> set, the proxy constructs a matching set, Ms. Initially, this set 
> contains all of the Accept-Contact predicates. Each of those predicates 
> is examined. It is matched to the contact predicate using the matching 
> operation of RFC 2533 [2]. If the result is not a match, and the 
> Accept-Contact predicate had its require flag set, the URI corresponding 
> to that contact predicate is discarded from the contact set. If the 
> result is not a match, but the Accept-Contact predicate removed from the 
> target set.

The above text is not actually whats in -08. The actual text is:

> Next, the proxy applies the predicates associated with the Accept-
>    Contact header field. For each contact that remains in the target
>    set, the proxy constructs a matching set, Ms. Initially, this set
>    contains all of the Accept-Contact predicates. Each of those
>    predicates is examined. It is matched to the contact predicate using
>    the matching operation of RFC 2533 [2]. If the result is not a match,
>    and the Accept-Contact predicate had its require flag set, the URI
>    corresponding to that contact predicate is discarded from the contact
>    set. If the result is not a match, but the Accept-Contact predicate
>    did not have its require flag set, that contact URI is not discarded
>    from the contact set, however, the Accept-Contact predicate is
>    removed from the matching set for that contact.


but your question is still relevant:

> 
> My question relates to the last statement in the above paragraph.
> The objective of that step, as I understand it, is to remove from the 
> set all predicates, that don't have an equivalent in the contact.
> 
> This step seems to be redundant, as it would not have an impact on the 
> score, since it would not change the score anyway (based on the way the 
> score is calculated)
> 
> Is my understanding correct?

In -08, I believe you are right. Setting the score to 0 is equivalent 
to removing the Accept-Contact predicate from the matching set. This 
algorithm has changed in -09, and with the new algorithm, it is not 
the case any longer.

> 
> My next question relates to the usage of the cancel-directive.
> Quoting the Draft :
> 
> cancel-directive: This type of directive indicates whether the
>              caller would like each proxy server to send a CANCEL
>              request downstream ("cancel") in response to a 200 OK from
>              the downstream server (which is the normal mode of
>              operation, making it somewhat redundant), or whether this
>              function should be left to the caller ("no-cancel"). If a
>              proxy receives a request with this parameter set to "no-
>              cancel", it SHOULD NOT CANCEL any outstanding branches on
>              receipt of a 2xx. However, it would still send CANCEL on
>              any outstanding branches on receipt of a 6xx.
> 
> The strength SHOULD NOT makes it unclear as to how the UAC can decide on 
> what to do, even if the proxy understands caller preferences.

The caller will end up seeing more 2xx than it normally would. It can 
then do what it wants - BYE some of them, keep some of them, as desired.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


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A New Internet-Draft is available from the on-line Internet-Drafts directories.
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	Title		: The Session Inititation Protocol (SIP) 'Join' Header
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	Filename	: draft-ietf-sip-join-02.txt
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This document defines a new header for use with SIP multi-party
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Henning,

 

Draft-ietf-sip-resource-priority-00.txt, Section 12, IANA Considerations,
asks not only for registration of the basic components of the RPH, i.e.,
namespace, ordered labels for priority, and a default, but adds that "the
registration MUST describe how SIP elements should treat requests from that
namespace, e.g., whether preemption or only preferential queuing are
allowed.".  This requirement really amounts to a service description of the
way the namespace is to be used.  

 

(As just a note, this position seems to be diametrically opposed to previous
discussions in ieprep (when the ID was under submission there) that seemed
to stress the protocol/policy distinction, in which the use of the namespace
for a service as a matter of policy.  However, I am glad to see the change.)

 

As I read the exchange between Mike Pierce and you on 24/25 June, you
concluded that assignments of namespaces should not be included in your RPH
ID.  I think that this is the right conclusion, particularly if the
namespace value assignments have to have an accompanying service
description.  

Am I correct in my understanding that you have agreed to take all the
namespace registrations our of your ID, i.e. all of Appendix B?

 

I would like to see the process conform to RFC 2434, "Guidelines for Writing
IANA Considerations Sections in RFCs",  which suggests that "One way to
insure community review of prospective assignments is to have the requester
submit a document for publication as an RFC".   We have a new ID,
http://www.ietf.org/internet-drafts/draft-mcgregor-ieprep-bridging-bcp-00.tx
t
<http://www.ietf.org/internet-drafts/draft-mcgregor-ieprep-bridging-bcp-00.t
xt> , that proposes a different structure to cover what you have in B.3 for
GETS.  If you are dropping Appendix B from your ID, we do not have to
further work that here.

 

But, it does raise the question, if there should be an RFC developed for a
RPH namespace to serve GETS, what working group should this be done under -
SIP, ieprep?

 

Dennis Berg

CSC

dennis.berg@dyncorp.com

 


------_=_NextPart_001_01C340B1.9B2ECAF0
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<div class=3DSection1>

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

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

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>Draft-ietf-sip-resource-priority-00.txt, Section 12, IANA
Considerations, asks not only for registration of the basic components =
of the
RPH, i.e., namespace, ordered labels for priority, and a default, but =
adds that
"the registration MUST describe how SIP elements should treat requests
from that namespace, e.g., whether preemption or only preferential =
queuing are
allowed.".&nbsp; This requirement really amounts to a service =
description
of the way the namespace is to be used.&nbsp; </span></font></p>

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

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>(As just a note, this position seems to be diametrically =
opposed to
previous discussions in ieprep (when the ID was under submission there) =
that
seemed to stress the protocol/policy distinction, in which the use of =
the
namespace for a service as a matter of policy.&nbsp; However, I am glad =
to see
the change.)</span></font></p>

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

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>As I read the exchange between Mike Pierce and you on 24/25 =
June, you
concluded that assignments of namespaces should not be included in your =
RPH
ID.&nbsp; I think that this is the right conclusion, particularly if =
the
namespace value assignments have to have an accompanying service
description.&nbsp; </span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>Am I correct in my understanding that you have agreed to take =
all the
namespace registrations our of your ID, i.e. all of Appendix =
B?</span></font></p>

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

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>I would like to see the process conform to RFC 2434, =
"Guidelines
for Writing IANA Considerations Sections in RFCs", &nbsp;which suggests
that "One way to insure community review of prospective assignments is =
to
have the requester submit a document for publication as an RFC".&nbsp; =
&nbsp;We
have a new ID, <a
href=3D"http://www.ietf.org/internet-drafts/draft-mcgregor-ieprep-bridgi=
ng-bcp-00.txt">http://www.ietf.org/internet-drafts/draft-mcgregor-ieprep=
-bridging-bcp-00.txt</a>,
that proposes a different structure to cover what you have in B.3 for
GETS.&nbsp; If you are dropping Appendix B from your ID, we do not have =
to
further work that here.</span></font></p>

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

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>But, it does raise the question, if there should be an RFC =
developed
for a RPH namespace to serve GETS, what working group should this be =
done under
- SIP, ieprep?</span></font></p>

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

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

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

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>dennis.berg@dyncorp.com</span></font></p>

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

</div>

</body>

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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		: SIP Authenticated Identity Body (AIB) Format
	Author(s)	: J. Peterson
	Filename	: draft-ietf-sip-authid-body-02.txt
	Pages		: 12
	Date		: 2003-7-2
	
RFC3261 introduces the concept of adding an S/MIME body to a SIP
request or response in order to provide reference integrity over its
headers.  This document provides a more specific mechanism to derive
integrity and authentication properties from an 'authenticated
identity body', a digitally-signed SIP message or message fragment.
A standard format for such bodies (known as Authenticated Identity
Bodies, or AIBs) is given in this document.  Some considerations for
the processing of AIBs by recipients of SIP messages with such bodies
are also given.

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	Title		: S/MIME AES Requirement for SIP
	Author(s)	: J. Peterson
	Filename	: draft-ietf-sip-smime-aes-01.txt
	Pages		: 6
	Date		: 2003-7-2
	
RFC3261 currently specifies 3DES as the required minimum ciphersuite
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normative guidance of RFC3261 to require the Advanced Encryption
Standard (AES) for S/MIME.

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

	Title		: Session Timers in the Session Initiation Protocol 
                          (SIP)
	Author(s)	: S. Donovan, J. Rosenberg
	Filename	: draft-ietf-sip-session-timer-11.txt
	Pages		: 21
	Date		: 2003-7-2
	
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, 
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From: Henning Schulzrinne <hgs@cs.columbia.edu>
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Subject: [Sip] Re: Namespace registration for SIP RPH
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Berg, Dennis wrote:

> but adds that "the registration MUST describe how SIP elements should 
> treat requests from that namespace, e.g., whether preemption or only 
> preferential queuing are allowed.".  This requirement really amounts to 
> a service description of the way the namespace is to be used. 

Correct.

> 
>  
> 
> (As just a note, this position seems to be diametrically opposed to 
> previous discussions in ieprep (when the ID was under submission there) 
> that seemed to stress the protocol/policy distinction, in which the use 
> of the namespace for a service as a matter of policy.  However, I am 
> glad to see the change.)

Not really. The notion was that the protocol element itself would not 
express handling procedures, i.e., something like

Resource-Handling: ;preemption=yes ;priority=strict

would be encoding policy in the element itself. The algorithm has to be 
described somewhere, after all, if this is to be useful. Clearly, a 
description can say "it is up to the implementation/local administrative 
policy/local drunk driving laws/today's lottery numbers whether 
preemption is allowed".

> 
>  
> 
> As I read the exchange between Mike Pierce and you on 24/25 June, you 
> concluded that assignments of namespaces should not be included in your 
> RPH ID.  I think that this is the right conclusion, particularly if the 
> namespace value assignments have to have an accompanying service 
> description. 

You must be referring to a different exchange of emails than I remember. 
The exchange was in reference to currently ill-defined namespaces, such 
as the ETS/TDR/etc. namespace. While I don't see this as a big issue 
(I'm happy to include namespaces or not), I was primarily concerned 
about any registrations that, due to external gating conditions, would 
delay the progress of the draft.

I would find it useful to include these initial registration if only to 
set an example of how we envision registrations to look like, while 
people are still paying attention to this.


> 
> Am I correct in my understanding that you have agreed to take all the 
> namespace registrations our of your ID, i.e. all of Appendix B?

I believe you misread our email exchange.

> 
>  
> 
> I would like to see the process conform to RFC 2434, "Guidelines for 
> Writing IANA Considerations Sections in RFCs",  which suggests that "One 
> way to insure community review of prospective assignments is to have the 
> requester submit a document for publication as an RFC".   We have a new 
> ID, 
> http://www.ietf.org/internet-drafts/draft-mcgregor-ieprep-bridging-bcp-00.txt, 
> that proposes a different structure to cover what you have in B.3 for 
> GETS.  If you are dropping Appendix B from your ID, we do not have to 
> further work that here.

As discussed, B.3 will be dropped since it is currently insufficiently 
well defined to be documented.

> 
>  
> 
> But, it does raise the question, if there should be an RFC developed for 
> a RPH namespace to serve GETS, what working group should this be done 
> under - SIP, ieprep?

I think the goal should be that namespace registrations primarily 
require domain expertise, not protocol expertise. Thus, the important 
part is that the constituency for the namespace is well-represented in 
the review.





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In a message dated 7/2/2003 11:52:47 AM Eastern Standard Time, 
Dennis.Berg@DynCorp.com writes (to Henning):


> Draft-ietf-sip-resource-priority-00.txt, Section 12, IANA Considerations, 
> asks not only for registration of the basic components of the RPH, i.e., 
> namespace, ordered labels for priority, and a default, but adds that "the 
> registration MUST describe how SIP elements should treat requests from that 
> namespace, e.g., whether preemption or only preferential queuing are allowed.".  This 
> requirement really amounts to a service description of the way the namespace 
> is to be used.  
>  
> (As just a note, this position seems to be diametrically opposed to previous 
> discussions in ieprep (when the ID was under submission there) that seemed 
> to stress the protocol/policy distinction, in which the use of the namespace 
> for a service as a matter of policy.  However, I am glad to see the change.)
>  
> As I read the exchange between Mike Pierce and you on 24/25 June, you 
> concluded that assignments of namespaces should not be included in your RPH ID.  I 
> think that this is the right conclusion, particularly if the namespace value 
> assignments have to have an accompanying service description.  
> Am I correct in my understanding that you have agreed to take all the 
> namespace registrations our of your ID, i.e. all of Appendix B?
> 


Dennis,

I agree that we should be very careful on how much we try to put in the R-P 
header document or require to be submitted with a registration with regards to 
procedures (preemption, preferential queuing, etc.) These can be described 
separately, and in fact, do not really need to be in any RFC or registration if 
they are local actions. I agree that the header definition should be limited to 
just that, but we've been pushed into describing some of the procedures for 
the use of the header.

My comments about definition of namespaces was to not include ones for ETS, 
GETS, TDR, 911, etc since there is a lot of uncertainty on what values were 
needed and how several namespaces would relate. On the other hand "Assured 
Service" (based on MLPP) is very clear so should be included.

Mike Pierce


--part1_175.1d12549f.2c34f736_boundary
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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 FAMILY=3D"SANSSERIF" FACE=
=3D"Arial" LANG=3D"0">In a message dated 7/2/2003 11:52:47 AM Eastern Standa=
rd Time, Dennis.Berg@DynCorp.com writes (to Henning):
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">Draft-ietf-sip-resource-pri=
ority-00.txt, Section 12, IANA Considerations, asks not only for registratio=
n of the basic components of the RPH, i.e., namespace, ordered labels for pr=
iority, and a default, but adds that "the registration MUST describe how SIP=
 elements should treat requests from that namespace, e.g., whether preemptio=
n or only preferential queuing are allowed.". &nbsp;This requirement really=20=
amounts to a service description of the way the namespace is to be used. &nb=
sp;</FONT><FONT  COLOR=3D"#000000" SIZE=3D2 FAMILY=3D"SERIF" FACE=3D"Times N=
ew Roman" LANG=3D"0">
<BR></FONT><FONT  COLOR=3D"#000000" SIZE=3D3 FAMILY=3D"SERIF" FACE=3D"Times=20=
New Roman" LANG=3D"0">=20
<BR>(As just a note, this position seems to be diametrically opposed to prev=
ious discussions in ieprep (when the ID was under submission there) that see=
med to stress the protocol/policy distinction, in which the use of the names=
pace for a service as a matter of policy. &nbsp;However, I am glad to see th=
e change.)
<BR>=20
<BR>As I read the exchange between Mike Pierce and you on 24/25 June, you co=
ncluded that assignments of namespaces should not be included in your RPH ID=
. &nbsp;I think that this is the right conclusion, particularly if the names=
pace value assignments have to have an accompanying service description. &nb=
sp;
<BR>Am I correct in my understanding that you have agreed to take all the na=
mespace registrations our of your ID, i.e. all of Appendix B?
<BR></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" SIZE=3D2 FAMILY=3D"SANSSERIF" FACE=3D"Ar=
ial" LANG=3D"0">
<BR>
<BR>Dennis,
<BR>
<BR>I agree that we should be very careful on how much we try to put in the=20=
R-P header document or require to be submitted with a registration with rega=
rds to procedures (preemption, preferential queuing, etc.) These can be desc=
ribed separately, and in fact, do not really need to be in any RFC or regist=
ration if they are local actions. I agree that the header definition should=20=
be limited to just that, but we've been pushed into describing some of the p=
rocedures for the use of the header.
<BR>
<BR>My comments about definition of namespaces was to not include ones for E=
TS, GETS, TDR, 911, etc since there is a lot of uncertainty on what values w=
ere needed and how several namespaces would relate. On the other hand "Assur=
ed Service" (based on MLPP) is very clear so should be included.
<BR>
<BR>Mike Pierce
<BR></FONT></HTML>

--part1_175.1d12549f.2c34f736_boundary--

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From exim@www1.ietf.org  Thu Jul  3 01:54:41 2003
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From: Rinchen Tundup <rinchen@cdotd.ernet.in>
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Subject: [Sip] Authorization header's nonce value
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Hello everyone,
 
I want to send credentials of a user in the first  REGISTER request ( in
Authorization header), but for computing the response (with MD5) i guess
we require the nonce value which is specified by the server in 401 or 407.

So my question is what should be the nonce value or it should be always 
first REGISTER with no credentials , 401 ( get nonce ) , REGISTER with
credentials , OK  or  am i missing
something?

Pls help . 

Greetings

Rinchen Tundup      
Research Engineer (Intelligent Networking)     
C-DOT Akbar Bhawan  Delhi 
24677525 Ext-102,312
  




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> 3. The definition of the Resource-Priority header should not allow 
> multiple resource values to be carried. I am unaware of any requirement 
> for this. The use described always needs exactly one value within one 
> namespace. Anything beyond this will bring on questions like "Which 
> priority has priority?" 

Mike,

I think I provided some of the motivation for this.

In the US PSTN, a call may be a GETS call AND a WPS call; a GETS call BUT
NOT a WPS call; or a WPS call BUT NOT a GETS call.

The "labels" for a GETS call and a WPS call are different (CPC parameter vs.
Precedence parameter)

Being a GETS call determines treatment in the landline network.  Being a WPS
call determines treatment in the wireless network.

It seems plausible that a similar situation could occur in  IP networks -
especially if one treatment was relevant in one administrative domain, but
another was available in a second administrative domain.

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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">
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5.5.2653.12">
<TITLE> multiple namespaces for SIP RPH</TITLE>
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<BR>

<P><FONT SIZE=3D2>&gt; 3. The definition of the Resource-Priority =
header should not allow </FONT>
<BR><FONT SIZE=3D2>&gt; multiple resource values to be carried. I am =
unaware of any requirement </FONT>
<BR><FONT SIZE=3D2>&gt; for this. The use described always needs =
exactly one value within one </FONT>
<BR><FONT SIZE=3D2>&gt; namespace. Anything beyond this will bring on =
questions like &quot;Which </FONT>
<BR><FONT SIZE=3D2>&gt; priority has priority?&quot; </FONT>
</P>

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

<P><FONT SIZE=3D2>I think I provided some of the motivation for =
this.</FONT>
</P>

<P><FONT SIZE=3D2>In the US PSTN, a call may be a GETS call AND a WPS =
call; a GETS call BUT NOT a WPS call; or a WPS call BUT NOT a GETS =
call.</FONT></P>

<P><FONT SIZE=3D2>The &quot;labels&quot; for a GETS call and a WPS call =
are different (CPC parameter vs. Precedence parameter)</FONT>
</P>

<P><FONT SIZE=3D2>Being a GETS call determines treatment in the =
landline network.&nbsp; Being a WPS call determines treatment in the =
wireless network.</FONT></P>

<P><FONT SIZE=3D2>It seems plausible that a similar situation could =
occur in&nbsp; IP networks - especially if one treatment was relevant =
in one administrative domain, but another was available in a second =
administrative domain.</FONT></P>

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My first post to this list so please be nice!
With the addition of "JOIN" does this make the total of valid methods 14?=
=A0=20
I have:

INVITE=A0 - RFC 2543
REGISTER - RFC 2543
BYE- RFC 2543
ACK- RFC 2543
CANCEL- RFC 2543
OPTIONS- RFC 2543

SUBSCRIBE - RFC 2848
UNSUBSCRIBE - RFC 2848
NOTIFY - RFC 2848

INFO - RFC 2976
PRACK - ??

UPDATE - RFC 3311
REFER - RFC 3515

JOIN -=A0 TBD

Ben Levitan
Raleigh, NC
919-782-7442
benlev@aol.com

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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 FAMILY=3D"SANSSERIF" FACE=
=3D"Arial" LANG=3D"0">My first post to this list so please be nice!<BR>
With the addition of "JOIN" does this make the total of valid methods 14?=
=A0 <BR>
I have:<BR>
<BR>
INVITE=A0 - RFC 2543<BR>
REGISTER - RFC 2543<BR>
BYE- RFC 2543<BR>
ACK- RFC 2543<BR>
CANCEL- RFC 2543<BR>
OPTIONS- RFC 2543<BR>
<BR>
SUBSCRIBE - RFC 2848<BR>
UNSUBSCRIBE - RFC 2848<BR>
NOTIFY - RFC 2848<BR>
<BR>
INFO - RFC 2976<BR>
PRACK - ??<BR>
<BR>
UPDATE - RFC 3311<BR>
REFER - RFC 3515<BR>
<BR>
JOIN -=A0 TBD<BR>
<BR>
Ben Levitan<BR>
Raleigh, NC<BR>
919-782-7442<BR>
benlev@aol.com<BR>
</FONT></HTML>
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Ben,
            Don't scare people on the list by using 2543.  RFC 3261 is
the current base line specification.
=20
Chris.
=20
=20
-----Original Message-----
From: BenLev@aol.com [mailto:BenLev@aol.com]=20
Sent: 03 July 2003 16:02
To: sip@ietf.org
Subject: [Sip] Methods inventory
=20
My first post to this list so please be nice!
With the addition of "JOIN" does this make the total of valid methods
14? =20
I have:

INVITE  - RFC 2543
REGISTER - RFC 2543
BYE- RFC 2543
ACK- RFC 2543
CANCEL- RFC 2543
OPTIONS- RFC 2543

SUBSCRIBE - RFC 2848
UNSUBSCRIBE - RFC 2848
NOTIFY - RFC 2848

INFO - RFC 2976
PRACK - ??

UPDATE - RFC 3311
REFER - RFC 3515

JOIN -  TBD

Ben Levitan
Raleigh, NC
919-782-7442
benlev@aol.com

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

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'><span =
style=3D'mso-tab-count:1'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp; </span>Don&#8217;t
scare people on the list by using 2543.<span =
style=3D'mso-spacerun:yes'>&nbsp; </span>RFC
3261 is the current base line =
specification.<o:p></o:p></span></font></p>

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

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

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

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

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0cm 4.0pt'>

<p class=3DMsoNormal><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;
font-family:Tahoma'>-----Original Message-----<br>
<b><span style=3D'font-weight:bold'>From:</span></b> BenLev@aol.com
[mailto:BenLev@aol.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> 03 July 2003 =
16:02<br>
<b><span style=3D'font-weight:bold'>To:</span></b> sip@ietf.org<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> [Sip] Methods =
inventory</span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>My first post to this list so please be nice!<br>
With the addition of &quot;JOIN&quot; does this make the total of valid =
methods
14?&nbsp; <br>
I have:<br>
<br>
INVITE&nbsp; - RFC 2543<br>
REGISTER - RFC 2543<br>
BYE- RFC 2543<br>
ACK- RFC 2543<br>
CANCEL- RFC 2543<br>
OPTIONS- RFC 2543<br>
<br>
SUBSCRIBE - RFC 2848<br>
UNSUBSCRIBE - RFC 2848<br>
NOTIFY - RFC 2848<br>
<br>
INFO - RFC 2976<br>
PRACK - ??<br>
<br>
UPDATE - RFC 3311<br>
REFER - RFC 3515<br>
<br>
JOIN -&nbsp; TBD<br>
<br>
Ben Levitan<br>
Raleigh, NC<br>
919-782-7442<br>
benlev@aol.com</span></font><o:p></o:p></p>

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Reference http://www.iana.org/assignments/sip-parameters for all SIP 
methods registered with IANA through standards track RFCs.

BenLev@aol.com wrote:

> My first post to this list so please be nice!
> With the addition of "JOIN" does this make the total of valid methods 
> 14? 
> I have:
>
> INVITE  - RFC 2543
> REGISTER - RFC 2543
> BYE- RFC 2543
> ACK- RFC 2543
> CANCEL- RFC 2543
> OPTIONS- RFC 2543
>
> SUBSCRIBE - RFC 2848
> UNSUBSCRIBE - RFC 2848
> NOTIFY - RFC 2848
>
> INFO - RFC 2976
> PRACK - ??
>
> UPDATE - RFC 3311
> REFER - RFC 3515
>
> JOIN -  TBD
>
> Ben Levitan
> Raleigh, NC
> 919-782-7442
> benlev@aol.com



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Reference <a href="http://www.iana.org/assignments/sip-parameters">http://www.iana.org/assignments/sip-parameters</a>
for all SIP methods registered with IANA through standards track RFCs.<br>
<br>
<a class="moz-txt-link-abbreviated" href="mailto:BenLev@aol.com">BenLev@aol.com</a> wrote:<br>
<blockquote type="cite" cite="midc.147f32b9.2c359f4f@aol.com"><font
 face="arial,helvetica"><font size="2" family="SANSSERIF" face="Arial"
 lang="0">My first post to this list so please be nice!<br>
 With the addition of "JOIN" does this make the total of valid methods 14?&nbsp;
  <br>
 I have:<br>
 <br>
 INVITE&nbsp; - RFC 2543<br>
 REGISTER - RFC 2543<br>
 BYE- RFC 2543<br>
 ACK- RFC 2543<br>
 CANCEL- RFC 2543<br>
 OPTIONS- RFC 2543<br>
 <br>
 SUBSCRIBE - RFC 2848<br>
 UNSUBSCRIBE - RFC 2848<br>
 NOTIFY - RFC 2848<br>
 <br>
 INFO - RFC 2976<br>
 PRACK - ??<br>
 <br>
 UPDATE - RFC 3311<br>
 REFER - RFC 3515<br>
 <br>
 JOIN -&nbsp; TBD<br>
 <br>
 Ben Levitan<br>
 Raleigh, NC<br>
 919-782-7442<br>
 <a class="moz-txt-link-abbreviated" href="mailto:benlev@aol.com">benlev@aol.com</a><br>
 </font></font></blockquote>
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Henning Schulzrinne wrote:
As discussed, B.3 will be dropped since it is currently insufficiently 
well defined to be documented.

[Berg, Dennis] Ok, we will then submit an ID to address the ETS namespace.
> 
Henning Schulzrinne wrote:
I think the goal should be that namespace registrations primarily 
require domain expertise, not protocol expertise. Thus, the important 
part is that the constituency for the namespace is well-represented in 
the review.
[Berg, Dennis] Fully agree.





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<P><FONT SIZE=2>Henning Schulzrinne wrote:</FONT>
<BR><FONT SIZE=2>As discussed, B.3 will be dropped since it is currently insufficiently </FONT>
<BR><FONT SIZE=2>well defined to be documented.</FONT>
</P>

<P><FONT SIZE=2>[Berg, Dennis] Ok, we will then submit an ID to address the ETS namespace.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>Henning Schulzrinne wrote:</FONT>
<BR><FONT SIZE=2>I think the goal should be that namespace registrations primarily </FONT>
<BR><FONT SIZE=2>require domain expertise, not protocol expertise. Thus, the important </FONT>
<BR><FONT SIZE=2>part is that the constituency for the namespace is well-represented in </FONT>
<BR><FONT SIZE=2>the review.</FONT>
<BR><FONT SIZE=2>[Berg, Dennis] Fully agree.</FONT>
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<BR>
<BR>

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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		: Caller Preferences for the Session Initiation Protocol
                          (SIP)
	Author(s)	: J. Rosenberg, H. Schulzrinne, P. Kyzivat
	Filename	: draft-ietf-sip-callerprefs-09.txt
	Pages		: 33
	Date		: 2003-7-3
	
This document describes a set of extensions to the Session Initiation
Protocol (SIP) which allow a caller to express preferences about
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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 four new request headers, Accept-Contact, Reject-
Contact, Require-Contact and Request-Disposition, which specify the
caller's preferences. The extension also defines new parameters for
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From exim@www1.ietf.org  Thu Jul  3 11:36:02 2003
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Subject: [Sip] about communication requirement.
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Hello,
I am interested in networked appliace control. So I started to examine
the requirements for networked appliances control. 
I read the draft-tsang-appliances-reqs-01.txt draft, but I could not
understand this requirement that is under "Communication Protocol
Requirements" category : 
 "   
   6.1  The communication protocol must provide a flexible payload 
   capability which will allow the transport of commands to, and 
   responses from, individual NAs or classes of NAs.  More explicitly, 
   there must be a separation of transport and data. 

 "
 I could not undertand what "seperation of transport and data" means?

 What I understood from these sentences is that, since the networked
appliaces have wide range of functionality, the payload must be flexible
enough to define the data or commands for each aplliance easily. is it
right?

 I know SIP is convenient for appliance control but at least could you
please tell me how can it fulfill this requirement?     

Thanks,

Ozge Guner
Anadolu University / TURKIYE

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Content-Type: text/plain

 

 

-----Original Message-----
From: Mpierce1@aol.com [mailto:Mpierce1@aol.com] 
Sent: Wednesday, July 02, 2003 11:04 PM
To: Berg, Dennis; hgs@cs.columbia.edu
Cc: sip@ietf.org; Gunn, Janet
Subject: Re: Namespace registration for SIP RPH

 

In a message dated 7/2/2003 11:52:47 AM Eastern Standard Time,
Dennis.Berg@DynCorp.com writes (to Henning): 





Draft-ietf-sip-resource-priority-00.txt, Section 12, IANA Considerations,
asks not only for registration of the basic components of the RPH, i.e.,
namespace, ordered labels for priority, and a default, but adds that "the
registration MUST describe how SIP elements should treat requests from that
namespace, e.g., whether preemption or only preferential queuing are
allowed.".  This requirement really amounts to a service description of the
way the namespace is to be used.   

(As just a note, this position seems to be diametrically opposed to previous
discussions in ieprep (when the ID was under submission there) that seemed
to stress the protocol/policy distinction, in which the use of the namespace
for a service as a matter of policy.  However, I am glad to see the change.)


As I read the exchange between Mike Pierce and you on 24/25 June, you
concluded that assignments of namespaces should not be included in your RPH
ID.  I think that this is the right conclusion, particularly if the
namespace value assignments have to have an accompanying service
description.   
Am I correct in my understanding that you have agreed to take all the
namespace registrations our of your ID, i.e. all of Appendix B? 




Dennis, 

I agree that we should be very careful on how much we try to put in the R-P
header document or require to be submitted with a registration with regards
to procedures (preemption, preferential queuing, etc.) These can be
described separately, and in fact, do not really need to be in any RFC or
registration if they are local actions. I agree that the header definition
should be limited to just that, but we've been pushed into describing some
of the procedures for the use of the header. 

My comments about definition of namespaces was to not include ones for ETS,
GETS, TDR, 911, etc since there is a lot of uncertainty on what values were
needed and how several namespaces would relate. On the other hand "Assured
Service" (based on MLPP) is very clear so should be included. 

Mike Pierce 

[Berg, Dennis] 

 

What document is then the "official" service description for the DSN
namespace?  Henning references RFC 791 in the B1  DSN namespace section.
That reference does not seem to be the most appropriate. James Polk in this
requirements references the ITU document (ITU-T Recommendation I.255.3
(1990), "Multilevel precedence and preemption service (MLPP)") which would
seem more relevant and complete.  


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

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

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma'>-----Original =
Message-----<br>
<b><span style=3D'font-weight:bold'>From:</span></b> Mpierce1@aol.com
[mailto:Mpierce1@aol.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> =
</span></font><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'>Wednesday, July
 02, 2003</span></font><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;
font-family:Tahoma'> </span></font><font size=3D2 face=3DTahoma><span
 style=3D'font-size:10.0pt;font-family:Tahoma'>11:04 =
PM</span></font><font
size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;font-family:Tahoma'><br>
<b><span style=3D'font-weight:bold'>To:</span></b> Berg, Dennis;
hgs@cs.columbia.edu<br>
<b><span style=3D'font-weight:bold'>Cc:</span></b> sip@ietf.org; Gunn, =
Janet<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> Re: Namespace
registration for SIP RPH</span></font></p>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 =
face=3D"Times New Roman"><span
style=3D'font-size:12.0pt'>&nbsp;</span></font></p>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial'>In a message dated =
</span></font><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>7/2/2003</span></font><font=

size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'> </span></font><font =
size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>11:52:47 =
AM</span></font><font
size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'> Eastern
Standard Time, Dennis.Berg@DynCorp.com writes (to Henning): <br>
<br>
<br>
<br>
</span></font></p>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D2 =
face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial'>Draft-ietf-sip-resource-pri=
ority-00.txt,
Section 12, IANA Considerations, asks not only for registration of the =
basic
components of the RPH, i.e., namespace, ordered labels for priority, =
and a
default, but adds that &quot;the registration MUST describe how SIP =
elements
should treat requests from that namespace, e.g., whether preemption or =
only
preferential queuing are allowed.&quot;. &nbsp;This requirement really =
amounts
to a service description of the way the namespace is to be used. =
&nbsp;</span></font><font
size=3D2 color=3Dblack FAMILY=3DSERIF><span =
style=3D'font-size:10.0pt;color:black'> <br>
</span></font><font color=3Dblack FAMILY=3DSERIF><span =
style=3D'color:black'><br>
(As just a note, this position seems to be diametrically opposed to =
previous
discussions in ieprep (when the ID was under submission there) that =
seemed to
stress the protocol/policy distinction, in which the use of the =
namespace for a
service as a matter of policy. &nbsp;However, I am glad to see the =
change.) <br>
<br>
As I read the exchange between Mike Pierce and you on 24/25 June, you =
concluded
that assignments of namespaces should not be included in your RPH ID. =
&nbsp;I
think that this is the right conclusion, particularly if the namespace =
value
assignments have to have an accompanying service description. &nbsp; =
<br>
Am I correct in my understanding that you have agreed to take all the =
namespace
registrations our of your ID, i.e. all of Appendix B? =
</span></font></p>

<p class=3DMsoNormal style=3D'margin-left:.5in'><font size=3D3 =
color=3Dblack
face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt;color:black'><br>
</span></font><font size=3D2 color=3Dblack face=3DArial =
FAMILY=3DSANSSERIF><span
style=3D'font-size:10.0pt;font-family:Arial;color:black'><br>
<br>
Dennis, <br>
<br>
I agree that we should be very careful on how much we try to put in the =
R-P
header document or require to be submitted with a registration with =
regards to
procedures (preemption, preferential queuing, etc.) These can be =
described
separately, and in fact, do not really need to be in any RFC or =
registration if
they are local actions. I agree that the header definition should be =
limited to
just that, but we've been pushed into describing some of the procedures =
for the
use of the header. <br>
<br>
My comments about definition of namespaces was to not include ones for =
ETS,
GETS, TDR, 911, etc since there is a lot of uncertainty on what values =
were
needed and how several namespaces would relate. On the other hand =
&quot;Assured
Service&quot; (based on MLPP) is very clear so should be included. <br>
<br>
Mike Pierce </span></font></p>

<p class=3DMsoNormal><font size=3D3 color=3Dblue face=3D"Times New =
Roman"><span
style=3D'font-size:12.0pt;color:blue'>[</span></font><font =
color=3Dblue><span
 style=3D'color:blue'>Berg, Dennis</span></font><font =
color=3Dblue><span
style=3D'color:blue'>] </span></font></p>

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

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D3 =
color=3Dblue
face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt;color:blue'>What document
is then the "official" service description for the DSN namespace? =
&nbsp;Henning
references RFC 791 in the B1 &nbsp;DSN namespace section.&nbsp; That =
reference
does not seem to be the most appropriate. James Polk in this =
requirements references
the ITU document (ITU-T Recommendation I.255.3 (1990), &quot;Multilevel
precedence and preemption service (MLPP</span></font>)&quot;<font =
color=3Dblue><span
style=3D'color:blue'>) which would seem more relevant and complete. =
&nbsp;</span></font></p>

</div>

</body>

</html>

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_______________________________________________
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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, H. Schulzrinne
	Filename	: draft-ietf-sip-symmetric-response-01.txt
	Pages		: 19
	Date		: 2003-7-3
	
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, and to the port written into the
topmost Via header field value of the request. This behavior is not
desirable in many cases, most notably, when the client is behind a
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From: "Brett Tate" <brett@broadsoft.com>
To: <sip@ietf.org>
Subject: RE: [Sip] Methods inventory
Date: Thu, 3 Jul 2003 13:42:20 -0400
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> My first post to this list so please be nice!

In case you are not aware,
sip-implementors@cs.columbia.edu should be used
for basic SIP questions.

http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors


> With the addition of "JOIN" does this make 
> the total of valid methods 14?

The following website contains information
about the registered sip methods and headers.

http://www.iana.org/assignments/sip-parameters

I'm unaware of an official public repository of 
all SIP methods within current or expired drafts.

However I guess that they could be culled from
http://www.softarmor.com/sipwg/


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In a message dated 7/3/2003 10:27:35 AM Eastern Standard Time, 
Janet.Gunn@DynCorp.com writes:


> In the US PSTN, a call may be a GETS call AND a WPS call; a GETS call BUT 
> NOT a WPS call; or a WPS call BUT NOT a GETS call.
> The "labels" for a GETS call and a WPS call are different (CPC parameter vs. 
> Precedence parameter) 
> Being a GETS call determines treatment in the landline network.  Being a WPS 
> call determines treatment in the wireless network.
> It seems plausible that a similar situation could occur in  IP networks - 
> especially if one treatment was relevant in one administrative domain, but 
> another was available in a second administrative domain.
> 

Yes, clearly a call can have multiple markings. It seems that we already have 
lots of possibilities, both in the circuit-switched world and IP. CPC, and 
its origins in SS#7 and US signaling, is one of them.

I just think that the Resource-Priority header and its purpose require that 
only one value (namespace) be carried in any single call setup attempt. That 
single value may then cause different actions or results for a wireless access, 
a wired access, an IP access link, a core link, a gateway, etc. The point is 
that the definition of the value needs to take into account all the 
requirements. If the Wireless access portion needs the 5 levels defined for WPS but GETS 
only needs one level while a regular call needs a "normal" level then we could 
define 6 levels. Once the call setup gets beyond the wireless access portion, 
the 5 individual levels defined for WPS may be treated the same. For 
assigning priority in a network, the level for GETS needs to be assigned relative to 
the WPS levels.

The R-P header should not be seen as a way to label the type of call (GETS, 
WPS, etc). It should just be a relative priority. If there is a need to label a 
call as WPS, or GETS, or something else, then there needs to be an explicit 
label for that (which allows either or both). The "namespace" should not be 
used for this.

Mike Pierce


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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2>In a message dated 7/3/20=
03 10:27:35 AM Eastern Standard Time, Janet.Gunn@DynCorp.com writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">In the US PSTN, a call may=20=
be a GETS call AND a WPS call; a GETS call BUT NOT a WPS call; or a WPS call=
 BUT NOT a GETS call.
<BR>The "labels" for a GETS call and a WPS call are different (CPC parameter=
 vs. Precedence parameter)=20
<BR>Being a GETS call determines treatment in the landline network. &nbsp;Be=
ing a WPS call determines treatment in the wireless network.
<BR>It seems plausible that a similar situation could occur in &nbsp;IP netw=
orks - especially if one treatment was relevant in one administrative domain=
, but another was available in a second administrative domain.
<BR></BLOCKQUOTE>
<BR>
<BR>Yes, clearly a call can have multiple markings. It seems that we already=
 have lots of possibilities, both in the circuit-switched world and IP. CPC,=
 and its origins in SS#7 and US signaling, is one of them.
<BR>
<BR>I just think that the Resource-Priority header and its purpose require t=
hat only one value (namespace) be carried in any single call setup attempt.=20=
That single value may then cause different actions or results for a wireless=
 access, a wired access, an IP access link, a core link, a gateway, etc. The=
 point is that the definition of the value needs to take into account all th=
e requirements. If the Wireless access portion needs the 5 levels defined fo=
r WPS but GETS only needs one level while a regular call needs a "normal" le=
vel then we could define 6 levels. Once the call setup gets beyond the wirel=
ess access portion, the 5 individual levels defined for WPS may be treated t=
he same. For assigning priority in a network, the level for GETS needs to be=
 assigned relative to the WPS levels.
<BR>
<BR>The R-P header should not be seen as a way to label the type of call (GE=
TS, WPS, etc). It should just be a relative priority. If there is a need to=20=
label a call as WPS, or GETS, or something else, then there needs to be an e=
xplicit label for that (which allows either or both). The "namespace" should=
 not be used for this.
<BR>
<BR>Mike Pierce
<BR></FONT></HTML>

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From exim@www1.ietf.org  Thu Jul  3 15:31:35 2003
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Date: Thu, 03 Jul 2003 15:29:40 -0400
From: Henning Schulzrinne <hgs@cs.columbia.edu>
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Subject: Re: [Sip] Re: multiple namespaces for SIP RPH
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Are you suggesting something that is essentially the cross-product of 
all priority namespaces? That seems rather unworkable.

There is no need to resolve the relative priority of A.X vs B.Y if they 
are not used by the same system (different administrations), i.e., they 
are orthogonal. If they are used by the same system, that system can 
presumably work out what it means if it receives a request containing 
both A.X and B.Y. This does not require standardization; in particular, 
there are any number of reasonable solutions, including a "virtual 
resource", so that the system sets aside some fraction for A.* and some 
fraction for B.*. It might do some internal ordering, but as long as the 
partial order with A.* and B.* is maintained, this doesn't seem to cause 
any difficulties.

I think of this as analogous to having both a 2nd class train ticket and 
a business class airfare - you need to carry both tickets with you as 
you travel, but the train conductor doesn't care about your airline 
ticket and vice versa.

Mpierce1@aol.com wrote:

> In a message dated 7/3/2003 10:27:35 AM Eastern Standard Time, 
> Janet.Gunn@DynCorp.com writes:
> 
> 
>> In the US PSTN, a call may be a GETS call AND a WPS call; a GETS call 
>> BUT NOT a WPS call; or a WPS call BUT NOT a GETS call.
>> The "labels" for a GETS call and a WPS call are different (CPC 
>> parameter vs. Precedence parameter)
>> Being a GETS call determines treatment in the landline network.  Being 
>> a WPS call determines treatment in the wireless network.
>> It seems plausible that a similar situation could occur in  IP 
>> networks - especially if one treatment was relevant in one 
>> administrative domain, but another was available in a second 
>> administrative domain.
> 
> 
> 
> Yes, clearly a call can have multiple markings. It seems that we already 
> have lots of possibilities, both in the circuit-switched world and IP. 
> CPC, and its origins in SS#7 and US signaling, is one of them.
> 
> I just think that the Resource-Priority header and its purpose require 
> that only one value (namespace) be carried in any single call setup 
> attempt. That single value may then cause different actions or results 
> for a wireless access, a wired access, an IP access link, a core link, a 
> gateway, etc. The point is that the definition of the value needs to 
> take into account all the requirements. If the Wireless access portion 
> needs the 5 levels defined for WPS but GETS only needs one level while a 
> regular call needs a "normal" level then we could define 6 levels. Once 
> the call setup gets beyond the wireless access portion, the 5 individual 
> levels defined for WPS may be treated the same. For assigning priority 
> in a network, the level for GETS needs to be assigned relative to the 
> WPS levels.
> 
> The R-P header should not be seen as a way to label the type of call 
> (GETS, WPS, etc). It should just be a relative priority. If there is a 
> need to label a call as WPS, or GETS, or something else, then there 
> needs to be an explicit label for that (which allows either or both). 
> The "namespace" should not be used for this.
> 
> Mike Pierce


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Subject: RE: [Sip] Re: multiple namespaces for SIP RPH
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Yes, that is a good analogy to what I was referring to. 

 And I want to make it clear that I am NOT interested in extending the
WPS/GETS "and/or" situation to IP domains- it is a "historical artifact".
Only that similar "artifacts" may arise in IP domains.

-----Original Message-----
From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
Sent: Thursday, July 03, 2003 3:30 PM

I think of this as analogous to having both a 2nd class train ticket and 
a business class airfare - you need to carry both tickets with you as 
you travel, but the train conductor doesn't care about your airline 
ticket and vice versa.

------_=_NextPart_001_01C3419A.05E63D90
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	charset="iso-8859-1"
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charset=3Diso-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
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<TITLE>RE: [Sip] Re: multiple namespaces for SIP RPH</TITLE>
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<P><FONT SIZE=3D2>Yes, that is a good analogy to what I was referring =
to. </FONT>
</P>

<P><FONT SIZE=3D2>&nbsp;And I want to make it clear that I am NOT =
interested in extending the WPS/GETS &quot;and/or&quot; situation to IP =
domains- it is a &quot;historical artifact&quot;.&nbsp; Only that =
similar &quot;artifacts&quot; may arise in IP domains.</FONT></P>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Henning Schulzrinne [<A =
HREF=3D"mailto:hgs@cs.columbia.edu">mailto:hgs@cs.columbia.edu</A>]</FON=
T>
<BR><FONT SIZE=3D2>Sent: Thursday, July 03, 2003 3:30 PM</FONT>
</P>

<P><FONT SIZE=3D2>I think of this as analogous to having both a 2nd =
class train ticket and </FONT>
<BR><FONT SIZE=3D2>a business class airfare - you need to carry both =
tickets with you as </FONT>
<BR><FONT SIZE=3D2>you travel, but the train conductor doesn't care =
about your airline </FONT>
<BR><FONT SIZE=3D2>ticket and vice versa.</FONT>
</P>

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In a message dated 7/3/2003 3:30:15 PM Eastern Standard Time, 
hgs@cs.columbia.edu writes:


> Are you suggesting something that is essentially the cross-product of 
> all priority namespaces? That seems rather unworkable.
> 
> There is no need to resolve the relative priority of A.X vs B.Y if they 
> are not used by the same system (different administrations), i.e., they 
> are orthogonal. If they are used by the same system, that system can 
> presumably work out what it means if it receives a request containing 
> both A.X and B.Y. This does not require standardization; in particular, 
> there are any number of reasonable solutions, including a "virtual 
> resource", so that the system sets aside some fraction for A.* and some 
> fraction for B.*. It might do some internal ordering, but as long as the 
> partial order with A.* and B.* is maintained, this doesn't seem to cause 
> any difficulties.
> 
> 


No, I am not suggesting any relationship between namespaces. Each is defined 
and registered separately. On the other hand, if a network chooses to handle 
multiple namespaces (one per call), it must decide how to determine the 
relative priorities when used on different calls that may contend for the same 
resources. There would be no standard and it would not be a part of the registration 
of each namespace.

But if there are multiple namespaces defined that are intended to be used in 
the same "domain" or environment or whatever you want to call it and on the 
same call, then I think it may become necessary to define the relative priority, 
maybe in registration. We need to avoid this.

The inclusion of multiple namespaces (or priorities) in a single SIP call 
setup request leads to the need for defining additional procedures to deal with 
them, including how to decide the sequence in which to try to interpret them.

>  think of this as analogous to having both a 2nd class train ticket and 
> a business class airfare - you need to carry both tickets with you as 
> you travel, but the train conductor doesn't care about your airline 
> ticket and vice versa.
> 

I love analogies. I think including two namespace.priority values in the SIP 
header is more analogous to having two airline tickets, one first class and 
one coach, handing both to the ticket agent, and expecting the agent to pick 
which one to use.

The airline/train analogy would be more like having a header for SIP to use 
and a separate parameter which only ISUP looks at. ISUP knows that it shouldn't 
look at the SIP header, just like the train conductor doesn't look at the 
airline ticket. This makes good sense.

Mike Pierce


--part1_ae.436fd492.2c35f38a_boundary
Content-Type: text/html; charset="US-ASCII"
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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2>In a message dated 7/3/20=
03 3:30:15 PM Eastern Standard Time, hgs@cs.columbia.edu writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">Are you suggesting somethin=
g that is essentially the cross-product of=20
<BR>all priority namespaces? That seems rather unworkable.
<BR>
<BR>There is no need to resolve the relative priority of A.X vs B.Y if they=20
<BR>are not used by the same system (different administrations), i.e., they=20
<BR>are orthogonal. If they are used by the same system, that system can=20
<BR>presumably work out what it means if it receives a request containing=20
<BR>both A.X and B.Y. This does not require standardization; in particular,=20
<BR>there are any number of reasonable solutions, including a "virtual=20
<BR>resource", so that the system sets aside some fraction for A.* and some=20
<BR>fraction for B.*. It might do some internal ordering, but as long as the=
=20
<BR>partial order with A.* and B.* is maintained, this doesn't seem to cause=
=20
<BR>any difficulties.
<BR>
<BR></FONT><FONT  COLOR=3D"#000000" SIZE=3D3 FAMILY=3D"SANSSERIF" FACE=3D"Ar=
ial" LANG=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" SIZE=3D2 FAMILY=3D"SANSSERIF" FACE=3D"Ar=
ial" LANG=3D"0">
<BR>
<BR>No, I am not suggesting any relationship between namespaces. Each is def=
ined and registered separately. On the other hand, if a network chooses to h=
andle multiple namespaces (one per call), it must decide how to determine th=
e relative priorities when used on different calls that may contend for the=20=
same resources. There would be no standard and it would not be a part of the=
 registration of each namespace.
<BR>
<BR>But if there are multiple namespaces defined that are intended to be use=
d in the same "domain" or environment or whatever you want to call it and on=
 the same call, then I think it may become necessary to define the relative=20=
priority, maybe in registration. We need to avoid this.
<BR>
<BR>The inclusion of multiple namespaces (or priorities) in a single SIP cal=
l setup request leads to the need for defining additional procedures to deal=
 with them, including how to decide the sequence in which to try to interpre=
t them.
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px"> think of this as analogous=
 to having both a 2nd class train ticket and=20
<BR>a business class airfare - you need to carry both tickets with you as=20
<BR>you travel, but the train conductor doesn't care about your airline=20
<BR>ticket and vice versa.
<BR></FONT><FONT  COLOR=3D"#000000" SIZE=3D3 FAMILY=3D"SANSSERIF" FACE=3D"Ar=
ial" LANG=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" SIZE=3D2 FAMILY=3D"SANSSERIF" FACE=3D"Ar=
ial" LANG=3D"0">
<BR>I love analogies. I think including two namespace.priority values in the=
 SIP header is more analogous to having two airline tickets, one first class=
 and one coach, handing both to the ticket agent, and expecting the agent to=
 pick which one to use.
<BR>
<BR>The airline/train analogy would be more like having a header for SIP to=20=
use and a separate parameter which only ISUP looks at. ISUP knows that it sh=
ouldn't look at the SIP header, just like the train conductor doesn't look a=
t the airline ticket. This makes good sense.
<BR>
<BR>Mike Pierce
<BR></FONT></HTML>

--part1_ae.436fd492.2c35f38a_boundary--

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Date: Thu, 03 Jul 2003 17:14:38 -0400
From: Henning Schulzrinne <hgs@cs.columbia.edu>
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CC: Janet.Gunn@DynCorp.com, sip@ietf.org, Dennis.Berg@DynCorp.com
Subject: Re: [Sip] Re: multiple namespaces for SIP RPH
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> 
> The inclusion of multiple namespaces (or priorities) in a single SIP 
> call setup request leads to the need for defining additional procedures 
> to deal with them, including how to decide the sequence in which to try 
> to interpret them.

Multiple priorities (A.X and A.Y) for the same namespace seem bad, but 
why would we have to define how a node handles

R-P: A.X, B.Y

? There are any number of reasonable options:

- Look for A; if A exists, ignore anything else.

- Look first for A, then B, then C, then .... Once found, stop and 
ignore the rest.

- Have some total order among the A and B priorities, defined strictly 
by local policy. Find A.X and B.Y and pick the "better" (or "worse") one.

Anything else?

If somebody really sees the need, we can't keep them from publishing 
such a total order ("If A and B are used in the same system, the 
following ordering ...."), but I don't see the need for this right now 
at the IETF standards level.

> 
>> think of this as analogous to having both a 2nd class train ticket and
>> a business class airfare - you need to carry both tickets with you as
>> you travel, but the train conductor doesn't care about your airline
>> ticket and vice versa.
> 
> 
> 
> I love analogies. I think including two namespace.priority values in the 
> SIP header is more analogous to having two airline tickets, one first 
> class and one coach, handing both to the ticket agent, and expecting the 
> agent to pick which one to use.

Not quite: Y and C tickets are from the same namespace. Think of having 
a Delta Gold/Silver Medaillon and a Business/Economy Ticket.

> 
> The airline/train analogy would be more like having a header for SIP to 
> use and a separate parameter which only ISUP looks at. ISUP knows that 
> it shouldn't look at the SIP header, just like the train conductor 
> doesn't look at the airline ticket. This makes good sense.

Again, a SIP request may visit any number of different administrations 
along the way to its destination, often unpredictable by the sender (due 
to name translations, among many other reasons).


> 
> Mike Pierce


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From exim@www1.ietf.org  Thu Jul  3 18:13:01 2003
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From: Dean Willis <dean.willis@softarmor.com>
To: sip@ietf.org
Cc: rohan@cisco.com
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Subject: [Sip] Initial agenda for SIP Working Group meeting at IETF 57
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Here's the initial agenda for SIP at IETF 57.

A pretty version with the reading-list hyperlinked in is available at: 
http://www.softarmor.com/sipwg/meets/ietf57/agenda.html

Send me any issues with this initial agenda. We'll work out what we can.
Of course, we'll probably make some last-minute adjustments, hence the
"agenda bash" that is always the first thing on the agenda.

--
Dean

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


Draft Agenda, SIP WG, IETF 57

________________________________________________________________________
Wednesay, July 16, 2003,  0900-1130 Hall IL

0900 Agenda Bash
Chairs

0905 Status of Work 
Chairs
IETF Last Calls 
draft-ietf-sip-content-indirect-mech-03.txt
draft-ietf-sip-sctp-03.txt
IESG Reviews
draft-ietf-sip-scvrtdisco-04.txt
draft-ietf-simple-event-list-04.txt
draft-ietf-sip-replaces-03.txt
draft-ietf-sip-symmetric-response-01.txt
WGLCs
draft-ietf-sip-authid-body-02.txt
draft-ietf-sip-referredby-02.txt

0930 Discussion of PUBLISH (WG)
Aki Niemi
draft-ietf-simple-publish-01.txt

0945 Discussion of Resource Priority (WG)
Henning Schulzrinne, James Polk
draft-ietf-sip-resource-priority-00.txt

0955 Discussion of Caller-prefs  (WG)
Jonathan Rosenberg
draft-ietf-sip-callerprefs-09.txt
draft-ietf-sipping-callerprefs-usecases-00.txt
draft-ietf-sip-callee-caps-00.txt

1010 Discussion of SIP Identity (WG)
Jon Peterson
draft-ietf-sip-asserted-identity-02.txt

1020 Discussion of SIP MIME (WG)
Jon Peterson
draft-ietf-sip-smime-aes-01.txt

1030 Discussion of History Info (WG)
Mary Barnes
draft-ietf-sip-history-info-00.txt

1040 Discussion of Sec-Inserted (Ind)
Mary Barnes
draft-barnes-sipping-sec-inserted-info-00.txt

1050 Discussion of Non-Invite transactions
Robert Sparks
draft-sparks-sip-noninvite-00.txt

1055 Discussion of Parameter Registry
Gonzalo Camarillo
draft-camarillo-sip-parameter-registry-01.txt
draft-camarillo-sip-uri-parameter-reg-00.txt

1105 Discussion of Connection Reuse
Rohan Mahy
draft-mahy-sip-connect-reuse-00.txt
draft-ietf-sipping-connect-reuse-reqs-00.txt

1115 Discussion of App Info
Cullen Jennings
draft-jennings-sip-app-info-01.txt

________________________________________________________________________


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From exim@www1.ietf.org  Sun Jul  6 09:02:00 2003
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From: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
To: <sip@ietf.org>
Cc: <klingle@cisco.com>, <jmaeng@ipdialog.com>, <drwalker@ss8.com>,
        <jf.mule@cablelabs.com>, "Bert Wijnen (E-mail)" <bwijnen@lucent.com>
Subject: [Sip] Comments on draft-ietf-sip-mib-06.txt
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Here are my comments concerning the latest version of the SIP MIB. I =
congratulate the authors for a much better document, and I thank them =
for taking into account and including fixes to most of the problems that =
were included in the review I did on the previous version.=20

There are still a few problems that need to be fixed:

A1) smilint complains about the following:

SIP-COMMON.txt:2014: [5] {integer-misuse} warning: use Integer32 instead =
of INTEGER in SMIv2 -- for sipStatusCodeValue --
SIP-COMMON.txt:2940: [5] {group-unref} warning: current group =
`sipCommonNotifObjectsGroup' is not referenced in this module

A2) You would like to re-consider the OID place holders that you created =
for sipRedirCfg, sipRedirStats, sipRegCfg, sipRegStats.
This is not an SMI violation, but it creates for the time being a hole =
in a MIB walk. If you need to add the definition of the respective =
objects in the future, you will not be able to do it in the same =
document, without the need to re-cycle the document at Proposed Standard =
level.=20

A3) DESCRIPTION Clause of sipServiceNotifEnable -  Saying that in case =
notifications are not implemented, the value has 'no meaning and MUST be =
0', seems contradictory and inaccurate. The respective sentence should =
read - 'The notifications are OPTIONAL, and if they are not implemented, =
this object's value MUST be all 0s.'

A4) How does sipNotifSequenceNumber work? It should probably be =
increased once for each notification defined in the SIP MIBs, but not =
for each notification target (in the case of multiple managers). I =
suggest this to be detailed in the DESCRIPTION clause for clarity.

=20

Minor editorial issues:

B1) English syntax in section 4.1. It should probably read 'The data =
type SipTransportProtocol is used as Textual Convention in this =
document. Textual conventions have...'
B2) Eliminate the commented text in the SIP compliance module of the UA =
MIB. =20

Regards,

Dan


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<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">Here are my comments =
concerning the latest version of the SIP MIB. I congratulate the authors =
for a much better document, and I thank them for taking into account and =
including fixes to most of the problems that were included in the review =
I did on the previous version. </FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">There are still a few =
problems that need to be fixed:</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">A1) smilint complains about =
the following:</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">SIP-COMMON.txt:2014: [5] =
{integer-misuse} warning: use Integer32 instead of INTEGER in SMIv2 -- =
for sipStatusCodeValue --</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">SIP-COMMON.txt:2940: [5] =
{group-unref} warning: current group `sipCommonNotifObjectsGroup' is not =
referenced in this module</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">A2) You would like to =
re-consider the OID place holders that you created for<SPAN =
LANG=3D"en-us"></SPAN></FONT><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"> <FONT FACE=3D"Times New Roman">sipRedirCfg, =
sipRedirStats, sipRegCfg, sipRegStats.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"he"><FONT SIZE=3D2 FACE=3D"Arial">This is not =
an SMI violation, but it creates for the time being a hole in a MIB =
walk. If you need to add the definition of the respective objects in the =
future, you will not be able to do it in the same document, without the =
need to re-cycle the document at Proposed Standard level. =
</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"he"><FONT SIZE=3D2 FACE=3D"Arial">A3) =
DESCRIPTION Clause of sipServiceNotifEnable -&nbsp; Saying that in case =
notifications are not implemented, the value has 'no meaning and MUST be =
0', seems contradictory and inaccurate. The respective sentence should =
read - 'The notifications are OPTIONAL, and if they are not implemented, =
this object's value MUST be all 0s.'</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"he"><FONT SIZE=3D2 FACE=3D"Arial">A4) How =
does sipNotifSequenceNumber work? It should probably be increased once =
for each notification defined in the SIP MIBs, but not for each =
notification target (in the case of multiple managers). I suggest this =
to be detailed in the DESCRIPTION clause for clarity.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"he"><FONT SIZE=3D2 =
FACE=3D"Arial">&nbsp;</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"he"><FONT SIZE=3D2 FACE=3D"Arial">Minor =
editorial issues:</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"he"><FONT SIZE=3D2 FACE=3D"Arial">B1) English =
syntax in section 4.1. It should probably read 'The data type =
SipTransportProtocol is used as Textual Convention in this document. =
Textual conventions have...'</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"he"><FONT SIZE=3D2 FACE=3D"Arial">B2) =
Eliminate the commented text in the SIP compliance module of the UA =
MIB.&nbsp; </FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"he"><FONT SIZE=3D2 =
FACE=3D"Arial">Regards,</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"he"><FONT SIZE=3D2 =
FACE=3D"Arial">Dan</FONT></SPAN></P>

</BODY>
</HTML>
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Date: Mon, 07 Jul 2003 16:49:18 +0900
From: "NAKAMURA, Hidefumi" <nakamura.hidefumi@lab.ntt.co.jp>
To: sip@ietf.org, mmusic@ietf.org
Cc: hidefumi.nakamura@staff.east.ntt.co.jp
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Subject: [Sip] Appropriate usage of SIP and RTSP
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Hi, all.

I have a question about the relationship between SIP and RTSP. 

In RFC3261, there is a description that says 
   "SIP is not a vertically integrated communications system.  SIP is
   rather a component that can be used with other IETF protocols to
   build a complete multimedia architecture.  Typically, these
   architectures will include protocols such as the Real-time Transport
   Protocol (RTP) (RFC 1889 [28]) for transporting real-time data and
   providing QoS feedback, the Real-Time streaming protocol (RTSP) (RFC
   2326 [29]) for controlling delivery of streaming media, ..."
   (RFC3261 Section 2)

However, my understanding is that both SIP and RTSP have the capability
for "discovery of UAs or media streams."
If I use SIP to establish a stream session and use RTSP to control
the stream, there will be redundant sequences, i.e. INVITE-OK-ACK and
SETUP-OK.

Was there any discussion on how to use these two protocols appropriately?

With SIP application server, network providers can provide various
services on contents delivery while we will not be able to do so only
with RTSP, I think.  On the other hand, RTSP has enough capability for
discovery of media streams and for negotiation of SDPs.
So, I think, for the simple delivery of contents, RTSP is appropriate to
use, and for various session control services regarding contents
delivery, SIP with RTSP is appropriate to use.

Any opinions?

I wonder why no contents delivery server seems to prepare SIP stack now. 
Is it because, up to now, there exists only a simple contents delivery
service?

best regards,

-- 
NAKAMURA, Hidefumi 
NTT ServiceIntegration /NetworkServiceSystems Labs.
tel: +81(422)59 3904; fax:+81(422)60 4012
---



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Date: Mon, 07 Jul 2003 18:28:19 +0900
From: "NAKAMURA, Hidefumi" <nakamura.hidefumi@lab.ntt.co.jp>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Cc: sip@ietf.org, mmusic@ietf.org
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Dear Magnus,

Thank you for your quick response.
I wrote down my opinion after your comments.

> Hi,
> 
> See comments inline:
> 
> NAKAMURA, Hidefumi wrote:
> > Hi, all.
> > 
> > I have a question about the relationship between SIP and RTSP. 
> > 
> > In RFC3261, there is a description that says 
> >    "SIP is not a vertically integrated communications system.  SIP is
> >    rather a component that can be used with other IETF protocols to
> >    build a complete multimedia architecture.  Typically, these
> >    architectures will include protocols such as the Real-time Transport
> >    Protocol (RTP) (RFC 1889 [28]) for transporting real-time data and
> >    providing QoS feedback, the Real-Time streaming protocol (RTSP) (RFC
> >    2326 [29]) for controlling delivery of streaming media, ..."
> >    (RFC3261 Section 2)
> > 
> > However, my understanding is that both SIP and RTSP have the capability
> > for "discovery of UAs or media streams."
> > If I use SIP to establish a stream session and use RTSP to control
> > the stream, there will be redundant sequences, i.e. INVITE-OK-ACK and
> > SETUP-OK.
> > 
> > Was there any discussion on how to use these two protocols appropriately?
> 
> Not to my knowledge. The SIP and RTSP combination has not been explored, 
> however it seem to contain a number of use cases. For example inviting a 
> media server into a SIP session or a conference session.
> 
> There are significant overlap in SIP and RTSP, mostly in regards to 
> session establishment that are done i different ways.
> > 
> > With SIP application server, network providers can provide various
> > services on contents delivery while we will not be able to do so only
> > with RTSP, I think.  On the other hand, RTSP has enough capability for
> > discovery of media streams and for negotiation of SDPs.
> > So, I think, for the simple delivery of contents, RTSP is appropriate to
> > use, and for various session control services regarding contents
> > delivery, SIP with RTSP is appropriate to use.
> 
> What do you mean with "session control services regarding contents 
> delivery"?

For example, if we establish a stream session via SIP application server
owned by the network provider, the application server can redirect the
session to the appropriate contents delivery server from the standpoint
of location, bandwidth prepared between terminals and the server, or
condition of contents server (i.e. in congestion, failure, and so on.)
Say, we get a media stream URI from a portal server.  The terminal(or
set top box) requests a session initiation to the media stream URI to
the SIP application server.  The SIP application server routes the
request to the contents delivery server that stores the media.  However,
if the server is in congestion and returns 503, the SIP application
server can select and redirect the session to another contents delivery
server instead.
I do not think this is possible with only RTSP.
(though the interaction among the portal server, contents delivery
servers and the terminals might enable this type of service in some
extent.)

> > Any opinions?
> > 
> > I wonder why no contents delivery server seems to prepare SIP stack now. 
> > Is it because, up to now, there exists only a simple contents delivery
> > service?
> 
> I believe that no one has looked into this and have had a need.

I understand.  Thank you very much for your comments.
Maybe it is interesting to investigate the use case, anyway.

> Best Regards
> 
> 
> Magnus Westerlund
> 
> Multimedia Technologies, Ericsson Research EAB/TVA/A
> ----------------------------------------------------------------------
> Ericsson AB                | Phone +46 8 4048287
> Torshamsgatan 23           | Fax   +46 8 7575550
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic

-- 
NAKAMURA, Hidefumi 
NTT ServiceIntegration /NetworkServiceSystems Labs.
tel: +81(422)59 3904; fax:+81(422)60 4012
---



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From exim@www1.ietf.org  Mon Jul  7 08:47:48 2003
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Subject: [Sip] Handling non-2xx response for BYE request
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Hi,


What is the correct behavior of a uac that receives a non-2xx resposne for a 
BYE request sent on an exisiting dialog ? should it terminate the call ?
According to rfc 3261 (15.1.1):
"A BYE request is constructed as would any other request within a dialog, as 
described in Section 12.
Once the BYE is constructed, the UAC core creates a new non-INVITE client 
transaction, and passes it the BYE request. The UAC MUST consider the 
session terminated (and therefore stop sending or listening for media) as 
soon as the BYE request is passed to the client transaction. If the response 
for the BYE is a 481 (Call/Transaction Does Not Exist) or a 408 (Request 
Timeout) or no response at all is received for the BYE (that is, a timeout 
is returned by the client transaction), the UAC MUST consider the session 
and the dialog terminated."


From the uas side -
what responses are reasonable for rejecting BYE, besides 401/407 and 481 ?
should it keep the dialog or terminate it after rejecting a BYE ?

if the uas reject the BYE but doesn't terminate the session, and the uac 
consider the session terminated as soon as it sent the BYE, then we might 
reach a-symetric situation where the uas has an active session but the uac 
already terminated the session.


thanks,

John.

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From: Marc Archer <marcher@aastra.com>
To: sip@ietf.org
Subject: RE: [Sip] I-D ACTION:draft-ietf-sip-mib-06.txt
Date: Mon, 7 Jul 2003 09:27:13 -0400 
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Some comments on the draft

TITLE: Has anyone come across a compilation error for not having a proper
IANA assigned number?

I am getting error during compilation (Sub-Id for item "sipTC" must be
"number" or "name (number)" format) as it is Draft version & we do not have
mib-2 number.  We got to have a number for SIP-COMMON-MIB, SIP-TC &
SIP-UA-MIB.

           ::= { mib-2 xx } 
   -- RFC Ed: replace xx with actual IANA assigned number  

Any thoughts? 

TITLE: Error due to BITS usage

I am experiencing a compilation error due to missing declaration of BITS in
SIP-COMMON-MIB & SIP-TC.

There is a BITS usage in the MIB files.

              SYNTAX     BITS {      
                               other(0),  -- none of the following     
                               udp(1),     
                               tcp(2),      
                               sctp(3),     
                               tls(4)     
              }      

In the IMPORTS Section, it was not defined; I was getting the following
Error message.

SMI item "BITS" used in SIP-TC, but not defined or imported.

Since "BITS" is defined in RFC1902.MIB (SNMPv2-SMI), which is already
included in the *.INC file, I added "BITS" in the IMPORTS Section as below:

      IMPORTS      
           MODULE-IDENTITY,    
           mib-2, BITS    
                FROM SNMPv2-SMI      

The compilation goes much further but it then comes up with the following
error:

expected a string, got BITS


Any help will be very much appreciated.

Regards,

Marc Archer

-----Original Message-----
From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
Sent: Wednesday, July 02, 2003 6:57 AM
Cc: sip@ietf.org
Subject: [Sip] I-D ACTION:draft-ietf-sip-mib-06.txt


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

	Title		: Management Information Base for Session Initiation

                          Protocol
	Author(s)	: K. Lingle, J. Maeng, J. Mule, D. Walker
	Filename	: draft-ietf-sip-mib-06.txt
	Pages		: 104
	Date		: 2003-7-1
	
This memo defines a portion of the Management Information Base (MIB) 
for use with network management protocols in the Internet community.  
In particular, it describes a set of managed objects that are used 
to manage Session Initiation Protocol (SIP) entities, which include 
User Agents, Proxy servers, Redirect servers and Registrars.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-mib-06.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-mib-06.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-mib-06.txt".
	
NOTE:	The mail server at ietf.org can return the document in
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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 exim@www1.ietf.org  Mon Jul  7 09:58:53 2003
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Subject: RE: [Sip] I-D ACTION:draft-ietf-sip-mib-06.txt
Date: Mon, 7 Jul 2003 16:57:34 +0300
Message-ID: <AAB4B3D3CF0F454F98272CBE187FDE2F04258D5F@is0004avexu1.global.avaya.com>
Thread-Topic: [Sip] I-D ACTION:draft-ietf-sip-mib-06.txt
Thread-Index: AcNEjcjm4ZAnf+CIQ9uZmvKoBwX/CwAAUFUA
From: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
To: "Marc Archer" <marcher@aastra.com>, <sip@ietf.org>
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Marc,

1) The OID assignment for the MIB modules is being done by RFC Editor in =
the final phases of releasing the RFC. Replace meantime xx with a number =
of your choice, trying to avoid the already defined numbers, in order to =
pass compilation.
2)  RFC 2578 Section 3.2 specifies which symbols must be imported and =
also lists certain pre-defined symbols that must not be imported. The =
BITS construct must not be imported.=20

For more information about standard MIBs review procedures (including =
compilation tools and options) see =
http://www.ietf.org/internet-drafts/draft-ietf-ops-mib-review-guidelines-=
01.txt.

Regards,

Dan
=20
> -----Original Message-----
> From: Marc Archer [mailto:marcher@aastra.com]
> Sent: 07 July, 2003 4:27 PM
> To: sip@ietf.org
> Subject: RE: [Sip] I-D ACTION:draft-ietf-sip-mib-06.txt
>=20
>=20
> Some comments on the draft
>=20
> TITLE: Has anyone come across a compilation error for not=20
> having a proper
> IANA assigned number?
>=20
> I am getting error during compilation (Sub-Id for item "sipTC" must be
> "number" or "name (number)" format) as it is Draft version &=20
> we do not have
> mib-2 number.  We got to have a number for SIP-COMMON-MIB, SIP-TC &
> SIP-UA-MIB.
>=20
>            ::=3D { mib-2 xx }=20
>    -- RFC Ed: replace xx with actual IANA assigned number =20
>=20
> Any thoughts?=20
>=20
> TITLE: Error due to BITS usage
>=20
> I am experiencing a compilation error due to missing=20
> declaration of BITS in
> SIP-COMMON-MIB & SIP-TC.
>=20
> There is a BITS usage in the MIB files.
>=20
>               SYNTAX     BITS {     =20
>                                other(0),  -- none of the=20
> following    =20
>                                udp(1),    =20
>                                tcp(2),     =20
>                                sctp(3),    =20
>                                tls(4)    =20
>               }     =20
>=20
> In the IMPORTS Section, it was not defined; I was getting the=20
> following
> Error message.
>=20
> SMI item "BITS" used in SIP-TC, but not defined or imported.
>=20
> Since "BITS" is defined in RFC1902.MIB (SNMPv2-SMI), which is already
> included in the *.INC file, I added "BITS" in the IMPORTS=20
> Section as below:
>=20
>       IMPORTS     =20
>            MODULE-IDENTITY,   =20
>            mib-2, BITS   =20
>                 FROM SNMPv2-SMI     =20
>=20
> The compilation goes much further but it then comes up with=20
> the following
> error:
>=20
> expected a string, got BITS
>=20
>=20
> Any help will be very much appreciated.
>=20
> Regards,
>=20
> Marc Archer
>=20
> -----Original Message-----
> From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
> Sent: Wednesday, July 02, 2003 6:57 AM
> Cc: sip@ietf.org
> Subject: [Sip] I-D ACTION:draft-ietf-sip-mib-06.txt
>=20
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> This draft is a work item of the Session Initiation Protocol=20
> Working Group
> of the IETF.
>=20
> 	Title		: Management Information Base for=20
> Session Initiation
>=20
>                           Protocol
> 	Author(s)	: K. Lingle, J. Maeng, J. Mule, D. Walker
> 	Filename	: draft-ietf-sip-mib-06.txt
> 	Pages		: 104
> 	Date		: 2003-7-1
> =09
> This memo defines a portion of the Management Information Base (MIB)=20
> for use with network management protocols in the Internet community. =20
> In particular, it describes a set of managed objects that are used=20
> to manage Session Initiation Protocol (SIP) entities, which include=20
> User Agents, Proxy servers, Redirect servers and Registrars.
>=20
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-sip-mib-06.txt
>=20
> To remove yourself from the IETF Announcement list, send a message to=20
> ietf-announce-request with the word unsubscribe in the body=20
> of the message.
>=20
> Internet-Drafts are also available by anonymous FTP. Login=20
> 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-mib-06.txt".
>=20
> A list of Internet-Drafts directories can be found in
> http://www.ietf.org/shadow.html=20
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>=20
>=20
> Internet-Drafts can also be obtained by e-mail.
>=20
> Send a message to:
> 	mailserv@ietf.org.
> In the body type:
> 	"FILE /internet-drafts/draft-ietf-sip-mib-06.txt".
> =09
> 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=20
> 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.
> 	=09
> 	=09
> Below is the data which will enable a MIME compliant mail reader
> implementation to automatically retrieve the ASCII version of the
> Internet-Draft.
>=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

_______________________________________________
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 exim@www1.ietf.org  Mon Jul  7 16:22:01 2003
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From: Marc Archer <marcher@aastra.com>
To: sip@ietf.org
Subject: RE: [Sip] I-D ACTION:draft-ietf-sip-mib-06.txt
Date: Mon, 7 Jul 2003 16:05:23 -0400 
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Hi,

One more query:

We are trying to integrate the Draft version for SIP MIBs (
http://www.ietf.org/internet-drafts/draft-ietf-sip-mib-06.txt). In this,
there is a BITS definition. BITS is not declared in the IMPORTS section (RFC
2578 Section 3.2) of the mib file. I have included RFC 1902 in the .INC file
as below:

#condInclude "rfc1902.inc" -- SNMPv2-SMI

When I compile, I am getting the error 

": f(sip_tc.mib), (3,1) SMI item "BITS" used in SIP-TC, but not defined or
imported".

I am using SMICng version 2.1.03(BOOK)(MS-DOS32), February 12, 1997.

Could someone tell me a fix for this problem? Is there any switch option
that can take care of this?

Regards,

Marc Archer

-----Original Message-----
From: Romascanu, Dan (Dan) [mailto:dromasca@avaya.com]
Sent: Monday, July 07, 2003 9:58 AM
To: Marc Archer; sip@ietf.org
Subject: RE: [Sip] I-D ACTION:draft-ietf-sip-mib-06.txt


Marc,

1) The OID assignment for the MIB modules is being done by RFC Editor in the
final phases of releasing the RFC. Replace meantime xx with a number of your
choice, trying to avoid the already defined numbers, in order to pass
compilation.
2)  RFC 2578 Section 3.2 specifies which symbols must be imported and also
lists certain pre-defined symbols that must not be imported. The BITS
construct must not be imported. 

For more information about standard MIBs review procedures (including
compilation tools and options) see
http://www.ietf.org/internet-drafts/draft-ietf-ops-mib-review-guidelines-01.
txt.

Regards,

Dan
 
> -----Original Message-----
> From: Marc Archer [mailto:marcher@aastra.com]
> Sent: 07 July, 2003 4:27 PM
> To: sip@ietf.org
> Subject: RE: [Sip] I-D ACTION:draft-ietf-sip-mib-06.txt
> 
> 
> Some comments on the draft
> 
> TITLE: Has anyone come across a compilation error for not 
> having a proper
> IANA assigned number?
> 
> I am getting error during compilation (Sub-Id for item "sipTC" must be
> "number" or "name (number)" format) as it is Draft version & 
> we do not have
> mib-2 number.  We got to have a number for SIP-COMMON-MIB, SIP-TC &
> SIP-UA-MIB.
> 
>            ::= { mib-2 xx } 
>    -- RFC Ed: replace xx with actual IANA assigned number  
> 
> Any thoughts? 
> 
> TITLE: Error due to BITS usage
> 
> I am experiencing a compilation error due to missing 
> declaration of BITS in
> SIP-COMMON-MIB & SIP-TC.
> 
> There is a BITS usage in the MIB files.
> 
>               SYNTAX     BITS {      
>                                other(0),  -- none of the 
> following     
>                                udp(1),     
>                                tcp(2),      
>                                sctp(3),     
>                                tls(4)     
>               }      
> 
> In the IMPORTS Section, it was not defined; I was getting the 
> following
> Error message.
> 
> SMI item "BITS" used in SIP-TC, but not defined or imported.
> 
> Since "BITS" is defined in RFC1902.MIB (SNMPv2-SMI), which is already
> included in the *.INC file, I added "BITS" in the IMPORTS 
> Section as below:
> 
>       IMPORTS      
>            MODULE-IDENTITY,    
>            mib-2, BITS    
>                 FROM SNMPv2-SMI      
> 
> The compilation goes much further but it then comes up with 
> the following
> error:
> 
> expected a string, got BITS
> 
> 
> Any help will be very much appreciated.
> 
> Regards,
> 
> Marc Archer
> 
> -----Original Message-----
> From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
> Sent: Wednesday, July 02, 2003 6:57 AM
> Cc: sip@ietf.org
> Subject: [Sip] I-D ACTION:draft-ietf-sip-mib-06.txt
> 
> 
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> This draft is a work item of the Session Initiation Protocol 
> Working Group
> of the IETF.
> 
> 	Title		: Management Information Base for 
> Session Initiation
> 
>                           Protocol
> 	Author(s)	: K. Lingle, J. Maeng, J. Mule, D. Walker
> 	Filename	: draft-ietf-sip-mib-06.txt
> 	Pages		: 104
> 	Date		: 2003-7-1
> 	
> This memo defines a portion of the Management Information Base (MIB) 
> for use with network management protocols in the Internet community.  
> In particular, it describes a set of managed objects that are used 
> to manage Session Initiation Protocol (SIP) entities, which include 
> User Agents, Proxy servers, Redirect servers and Registrars.
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-sip-mib-06.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-mib-06.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-mib-06.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.
> 
> _______________________________________________
> 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: Kevin Lingle <klingle@cisco.com>
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To: Marc Archer <marcher@aastra.com>
CC: sip@ietf.org
Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-mib-06.txt
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marc,

we (cisco) used to have essentially the same problem due to
older verison of smicng not recognizing BITS construct.
i believe newer versions of smicng (i think we have v2.2.11 now)
have fixed this afaik.   i know that we used to have to "work around"
this by commenting out BITS syntax and replacing it with OCTET STRING.

i suggest you contact http://www.snmpinfo.com for a definitive answer on
whether your version of smicng is deficient wrt BITS or whether
there is a switch you can turn on to eliminate the compile problem.

btw, i ran the mib modules though smicng before publishing and did
not have any errors reported.... except for the mib numbers: xx, yy
which are eliminated by putting in some arbitrary (high) mib-2 oid
assigned numbers (per dan's earlier suggestion).

kevin

Marc Archer wrote:
> Hi,
> 
> One more query:
> 
> We are trying to integrate the Draft version for SIP MIBs (
> http://www.ietf.org/internet-drafts/draft-ietf-sip-mib-06.txt). In this,
> there is a BITS definition. BITS is not declared in the IMPORTS section (RFC
> 2578 Section 3.2) of the mib file. I have included RFC 1902 in the .INC file
> as below:
> 
> #condInclude "rfc1902.inc" -- SNMPv2-SMI
> 
> When I compile, I am getting the error 
> 
> ": f(sip_tc.mib), (3,1) SMI item "BITS" used in SIP-TC, but not defined or
> imported".
> 
> I am using SMICng version 2.1.03(BOOK)(MS-DOS32), February 12, 1997.
> 
> Could someone tell me a fix for this problem? Is there any switch option
> that can take care of this?
> 
> Regards,
> 
> Marc Archer
> 
> -----Original Message-----
> From: Romascanu, Dan (Dan) [mailto:dromasca@avaya.com]
> Sent: Monday, July 07, 2003 9:58 AM
> To: Marc Archer; sip@ietf.org
> Subject: RE: [Sip] I-D ACTION:draft-ietf-sip-mib-06.txt
> 
> 
> Marc,
> 
> 1) The OID assignment for the MIB modules is being done by RFC Editor in the
> final phases of releasing the RFC. Replace meantime xx with a number of your
> choice, trying to avoid the already defined numbers, in order to pass
> compilation.
> 2)  RFC 2578 Section 3.2 specifies which symbols must be imported and also
> lists certain pre-defined symbols that must not be imported. The BITS
> construct must not be imported. 
> 
> For more information about standard MIBs review procedures (including
> compilation tools and options) see
> http://www.ietf.org/internet-drafts/draft-ietf-ops-mib-review-guidelines-01.
> txt.
> 
> Regards,
> 
> Dan
>  
> 
>>-----Original Message-----
>>From: Marc Archer [mailto:marcher@aastra.com]
>>Sent: 07 July, 2003 4:27 PM
>>To: sip@ietf.org
>>Subject: RE: [Sip] I-D ACTION:draft-ietf-sip-mib-06.txt
>>
>>
>>Some comments on the draft
>>
>>TITLE: Has anyone come across a compilation error for not 
>>having a proper
>>IANA assigned number?
>>
>>I am getting error during compilation (Sub-Id for item "sipTC" must be
>>"number" or "name (number)" format) as it is Draft version & 
>>we do not have
>>mib-2 number.  We got to have a number for SIP-COMMON-MIB, SIP-TC &
>>SIP-UA-MIB.
>>
>>           ::= { mib-2 xx } 
>>   -- RFC Ed: replace xx with actual IANA assigned number  
>>
>>Any thoughts? 
>>
>>TITLE: Error due to BITS usage
>>
>>I am experiencing a compilation error due to missing 
>>declaration of BITS in
>>SIP-COMMON-MIB & SIP-TC.
>>
>>There is a BITS usage in the MIB files.
>>
>>              SYNTAX     BITS {      
>>                               other(0),  -- none of the 
>>following     
>>                               udp(1),     
>>                               tcp(2),      
>>                               sctp(3),     
>>                               tls(4)     
>>              }      
>>
>>In the IMPORTS Section, it was not defined; I was getting the 
>>following
>>Error message.
>>
>>SMI item "BITS" used in SIP-TC, but not defined or imported.
>>
>>Since "BITS" is defined in RFC1902.MIB (SNMPv2-SMI), which is already
>>included in the *.INC file, I added "BITS" in the IMPORTS 
>>Section as below:
>>
>>      IMPORTS      
>>           MODULE-IDENTITY,    
>>           mib-2, BITS    
>>                FROM SNMPv2-SMI      
>>
>>The compilation goes much further but it then comes up with 
>>the following
>>error:
>>
>>expected a string, got BITS
>>
>>
>>Any help will be very much appreciated.
>>
>>Regards,
>>
>>Marc Archer
>>
>>-----Original Message-----
>>From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
>>Sent: Wednesday, July 02, 2003 6:57 AM
>>Cc: sip@ietf.org
>>Subject: [Sip] I-D ACTION:draft-ietf-sip-mib-06.txt
>>
>>
>>A New Internet-Draft is available from the on-line Internet-Drafts
>>directories.
>>This draft is a work item of the Session Initiation Protocol 
>>Working Group
>>of the IETF.
>>
>>	Title		: Management Information Base for 
>>Session Initiation
>>
>>                          Protocol
>>	Author(s)	: K. Lingle, J. Maeng, J. Mule, D. Walker
>>	Filename	: draft-ietf-sip-mib-06.txt
>>	Pages		: 104
>>	Date		: 2003-7-1
>>	
>>This memo defines a portion of the Management Information Base (MIB) 
>>for use with network management protocols in the Internet community.  
>>In particular, it describes a set of managed objects that are used 
>>to manage Session Initiation Protocol (SIP) entities, which include 
>>User Agents, Proxy servers, Redirect servers and Registrars.
>>
>>A URL for this Internet-Draft is:
>>http://www.ietf.org/internet-drafts/draft-ietf-sip-mib-06.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-mib-06.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-mib-06.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.
>>
>>_______________________________________________
>>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
> 


-- 
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
  Kevin R. Lingle       919.392.2029
  http://www.klove.com                               http://www.air1.com
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=


_______________________________________________
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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 exim@www1.ietf.org  Tue Jul  8 02:55:43 2003
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Message-ID: <3F0A6A8D.3040903@dynamicsoft.com>
Date: Tue, 08 Jul 2003 02:54:05 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Subject: [Sip] New I-D on callee capabilities
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Folks,

You may have noticed that I posted a new I-D on SIP callee capabilities:

http://www.ietf.org/internet-drafts/draft-ietf-sip-callee-caps-00.txt
http://www.jdrosen.net/papers/draft-ietf-sip-callee-caps-00.txt
http://www.jdrosen.net/papers/draft-ietf-sip-callee-caps-00.html

This represents one half of the split of the former caller preferences 
spec. The above draft considers only capabilities - setting the 
contact parameters in REGISTER, OPTIONS, target refresh requests.
Functionally, there is no change from the capabilities aspects of 
caller-prefs-08. The document is substantially reorganized, however, 
and I think reads pretty well.

The reason for the split was that we believed (and I still do 
believe), that we are going to continue to have a tough time with the 
caller preferences piece. However, several key documents are blocked 
on a reference to callerprefs. By doing this split, we can move 
forward quickly with the capabilities doc. Most of the current docs 
blocked on a reference are actually blocked on the capabilities piece. 
  So, they will be able to proceed as well.

Comments/questions welcome as always.

Thanks,
Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com



_______________________________________________
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Date: Tue, 08 Jul 2003 03:15:31 -0400
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Subject: [Sip] Changes in caller prefs -09
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Folks,

You may have noticed that caller prefs -09 was posted:

http://www.jdrosen.net/papers/draft-ietf-sip-callerprefs-09.txt
http://www.jdrosen.net/papers/draft-ietf-sip-callerprefs-09.html
http://www.ietf.org/internet-drafts/draft-ietf-sip-callerprefs-09.txt

This document is substantially different from -08.

First, the capabilities stuff has been yanked into a separate I-D 
(http://www.ietf.org/internet-drafts/draft-ietf-sip-callee-caps-00.txt). 
That includes REGISTER and OPTIONS processing, and all of the media 
feature tag definitions and registrations.

The reason for the split, as I indicated in another note, was that we 
believed there were going to continue to be problems with the caller 
preferences piece of callerprefs (i.e., the Accept-Contact and 
Reject-Contact header fields). A big issue we encountered with -08 
based on excellent input from Ted Hardie, was that q-value arithmetic 
is a no-no. Q-values as defined are ordinal, and you can't perform 
arithmetic on them, or compare them between different sources. We had 
been doing a lot of that in -08.

So, in -09, the algorithm has changed. THe proxy computes the score 
for each Accept-Contact rule against each Contact, as before. The 
overall caller preference for a contact is then computed as the 
average of the scores across all Accept-Contacts for that contact. 
Previously, we had a really complex function instead of arithmetic 
average. The reason had a lot to do with quirks that arise when you 
try to do q-value arithmetic. Anyway, once the average (the caller 
preference) is computed, it is used in a specific way. For those 
contacts with EQUAL Q-VALUES AS SET BY THE CALLEE, the caller 
preference is used to order them. We never combine the caller and 
callee preferences; we merely use the caller preference to provide an 
order when the callee provides none.

This algorithm is a lot simpler, and it never tries to do arithmetic 
on q-values. However, there is some loss of functionality. I tested 
the algorithm on all of the use cases, and found that a few failed. 
For example, Section 3.5 of 
http://www.jdrosen.net/papers/draft-ietf-sipping-callerprefs-usecases-00.txt 
describes a case. There, the caller wants the call to go to a 
videophone, but will take an audio-only device as a second choice. The 
callee prefers the audio-only phone. Because the caller preferences 
can't ever reorder callee contacts with differing q-values, there is 
no way to "override" the callee preference for the audio phone, and 
still fall back to it in the case where the videophone doesnt work or 
can't be reached.

Section 3.12 describes a similar case, where a user wants to go right 
to voicemail, but will take a call with the user if there is no 
voicemail. We can't do that anymore.

I welcome comments on whether this loss of functionality is 
substantial enough that we need to investigate ways of getting it 
back. I personally would prefer to just declare victory with what 
works and move on.

Some of the other changes:

* elimination of the q-values in the Accept-Contact; they were never 
needed in any of the use cases, so I removed them

* Clarification on the purpose of Proxy-Require, based on comments by 
Mary and others during wglc.

* A lot more discussion on how to set the require and explicit 
parameters. These can be confusing, and its hard to tell which of them 
you need.

* when a proxy is redirecting, it sets the q-values for the contacts 
to anything which orders them based on the results of the above algorithm


Comments and questions welcome.

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com



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From exim@www1.ietf.org  Tue Jul  8 11:33:51 2003
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From: The IESG <iesg-secretary@ietf.org>
Date: Tue, 08 Jul 2003 11:31:59 -0400
Subject: [Sip] Protocol Action: An Extension to the Session Initiation
 Protocol (SIP)  for Symmetric Response Routing to Proposed
 Standard
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The IESG has approved the Internet-Draft 'An Extension to the Session
Initiation Protocol (SIP) for Symmetric Response Routing'
<draft-ietf-sip-symmetric-response-01.txt> as a Proposed Standard.
This document is the product of the Session Initiation Protocol
Working Group. The IESG contact persons are Allison Mankin and Jon
Peterson.

Technical Summary

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, and to the port written into the
topmost Via header field value of the request. This behavior is not
desirable in many cases, most notably, when the client is behind a
Network Address Translator (NAT). This extension defines a new
parameter for the Via header field, 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.

Working Group Summary

The Working Group supported the advancement of this document.
There were reviews by working group members.

Protocol Quality

The document includes an analysis of the work in the perspective
of the IAB's UNSAF Considerations. The review for the IESG was
carried out by Allison Mankin.


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From exim@www1.ietf.org  Tue Jul  8 16:42:56 2003
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Date: Tue, 08 Jul 2003 16:40:20 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: "NAKAMURA, Hidefumi" <nakamura.hidefumi@lab.ntt.co.jp>
CC: Magnus Westerlund <magnus.westerlund@ericsson.com>, sip@ietf.org,
        mmusic@ietf.org
References: <20030707134023.181C.NAKAMURA.HIDEFUMI@lab.ntt.co.jp> <3F0930B4.8090202@ericsson.com> <20030707174715.1826.NAKAMURA.HIDEFUMI@lab.ntt.co.jp>
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Subject: [Sip] Re: [MMUSIC] Appropriate usage of SIP and RTSP
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I agree that there is significant overlap between RTSP and SIP, and 
this is something that has been bothering more and more over time. I 
have had in mind to write a draft on the ways in which they could be 
better merged, but have not had the time.

The primary place where they overlap is session establishment. In this 
area, I think SIP is much richer, in terms of features. SIP would 
allow connections to media servers running on PCs and other end-user 
devices. Its rich routing facilities could allow nice policy-based 
connection to content. It has had a lot of work on firewall/nat 
traversal which is now only getting repeated for RTSP. Plus, in many 
cases, you might initiate a session, but not know, in advance, that 
the session is to be connected to a content server. As an example, I'd 
like to be able to make a sip call to a user, and instead get 
connected to a voicemail greeting that I can control with RTSP.

IMHO, the most obvious way to make this work is to use SIP to initiate 
a session. If the session is to a media server, then, using SDP 
negotiations, one can add an additional "media channel", which is 
actually RTSP, doing just media controls (play, fast-forward, etc.). 
Thus, SIP would set up an RTSP session to control the media that the 
SIP call connected to.

I havent really fleshed this out much, but it seemed like a start.

-Jonathan R.

NAKAMURA, Hidefumi wrote:

> Dear Magnus,
> 
> Thank you for your quick response.
> I wrote down my opinion after your comments.
> 
> 
>>Hi,
>>
>>See comments inline:
>>
>>NAKAMURA, Hidefumi wrote:
>>
>>>Hi, all.
>>>
>>>I have a question about the relationship between SIP and RTSP. 
>>>
>>>In RFC3261, there is a description that says 
>>>   "SIP is not a vertically integrated communications system.  SIP is
>>>   rather a component that can be used with other IETF protocols to
>>>   build a complete multimedia architecture.  Typically, these
>>>   architectures will include protocols such as the Real-time Transport
>>>   Protocol (RTP) (RFC 1889 [28]) for transporting real-time data and
>>>   providing QoS feedback, the Real-Time streaming protocol (RTSP) (RFC
>>>   2326 [29]) for controlling delivery of streaming media, ..."
>>>   (RFC3261 Section 2)
>>>
>>>However, my understanding is that both SIP and RTSP have the capability
>>>for "discovery of UAs or media streams."
>>>If I use SIP to establish a stream session and use RTSP to control
>>>the stream, there will be redundant sequences, i.e. INVITE-OK-ACK and
>>>SETUP-OK.
>>>
>>>Was there any discussion on how to use these two protocols appropriately?
>>
>>Not to my knowledge. The SIP and RTSP combination has not been explored, 
>>however it seem to contain a number of use cases. For example inviting a 
>>media server into a SIP session or a conference session.
>>
>>There are significant overlap in SIP and RTSP, mostly in regards to 
>>session establishment that are done i different ways.
>>
>>>With SIP application server, network providers can provide various
>>>services on contents delivery while we will not be able to do so only
>>>with RTSP, I think.  On the other hand, RTSP has enough capability for
>>>discovery of media streams and for negotiation of SDPs.
>>>So, I think, for the simple delivery of contents, RTSP is appropriate to
>>>use, and for various session control services regarding contents
>>>delivery, SIP with RTSP is appropriate to use.
>>
>>What do you mean with "session control services regarding contents 
>>delivery"?
> 
> 
> For example, if we establish a stream session via SIP application server
> owned by the network provider, the application server can redirect the
> session to the appropriate contents delivery server from the standpoint
> of location, bandwidth prepared between terminals and the server, or
> condition of contents server (i.e. in congestion, failure, and so on.)
> Say, we get a media stream URI from a portal server.  The terminal(or
> set top box) requests a session initiation to the media stream URI to
> the SIP application server.  The SIP application server routes the
> request to the contents delivery server that stores the media.  However,
> if the server is in congestion and returns 503, the SIP application
> server can select and redirect the session to another contents delivery
> server instead.
> I do not think this is possible with only RTSP.
> (though the interaction among the portal server, contents delivery
> servers and the terminals might enable this type of service in some
> extent.)
> 
> 
>>>Any opinions?
>>>
>>>I wonder why no contents delivery server seems to prepare SIP stack now. 
>>>Is it because, up to now, there exists only a simple contents delivery
>>>service?
>>
>>I believe that no one has looked into this and have had a need.
> 
> 
> I understand.  Thank you very much for your comments.
> Maybe it is interesting to investigate the use case, anyway.
> 
> 
>>Best Regards
>>
>>
>>Magnus Westerlund
>>
>>Multimedia Technologies, Ericsson Research EAB/TVA/A
>>----------------------------------------------------------------------
>>Ericsson AB                | Phone +46 8 4048287
>>Torshamsgatan 23           | Fax   +46 8 7575550
>>S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>>
>>
>>_______________________________________________
>>mmusic mailing list
>>mmusic@ietf.org
>>https://www1.ietf.org/mailman/listinfo/mmusic
> 
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


_______________________________________________
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------=_NextPart_000_0025_01C34605.7471C620--


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From exim@www1.ietf.org  Wed Jul  9 03:10:45 2003
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Subject: [Sip] question about mime mmutipart mixed, in case of SIP-T
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  Hello,

  I have a simple question about multipart bodies...

  Suppose the following message:

S I  P /  2 .  0    1 8  0    R i  n g  i n  g
 M  I M  E -  v e  r s  i o  n :  1 .  0
 C  o n  t e  n t  - T  y p  e :    m u  l t  i p  a r  t /  m i  x e  d ;b
o  u n  d a  r y  = b  o u  n d  a r  y1
 T  o :  < s i p  : 0  2 1  4 4  4 4  1 0  4 1  @ 2 1 8  . 1  . 1  3 0  . 7
1 ;  U s  e r= p  h o  n e  > ;  t a  g =  0 0  0 00 0  0 0  0 0  0 0  6 6
A F  8 9  5 50 D  0 0
 F r  o m  : <  s i  p : 0 5  7 1  8 6  5 4  1 2  0 2  @ 6  1 .1 3  0 .  1
.  2 0  ; U  s e  r =  p ho n  e >  ; t  a g  = 3  f 0  3 f  e 52 -  2 3  c
a  - 3  d 8  2 0  1 1  4
 V  i a  : S  I P  / 2  . 0  / U  D P  6  1 .  1 3  0 .  1 .  2 0  : 5  0 6
0 ;  b r  a n  c h  = z  9 h  G 4  b K 1 2  4 5  6 6  1 3  5 1  .
C  a l   l -  I D  : 3  F 0  3 F  E 5  2 -  0 0  0 0  0 0  0 0  @ J  F S  I
P  P C  U 1
C S  e q  : 1    I  N V  I T  E
R  e q  u i  r e  : 1  0 0  r e  l
R  S e  q :  4 3  9 8  1
C  o n  t a  c t  : <  s i  p :  0 2  1 4  4 4  4 1  0 4  1 @  2 1  8 .  1
.  1 3  0 . 7 1  : 5  0 6  0 >
D a  t e  : T  h u  ,    3    J u  l    2 0  0 3    1 7 :  2 8  : 0  9    G
M  T
T  i m  e s  t a  m p  : 2  1 8  9 7
S u  p p  o r  t e  d :  1 0  0 r  e l
C o  n t  e n  t -  L e  n g  t h  : 2 4 9

  - -  b o  u n  d a  r y1
C o  n t  e n t -  T y  p e  : a  p p  l i  c a  t i o n  / S  D P

 v =  0
o  = -    1  2    1 2    I  N    I P  4  2 1  8 .  1 .  1 3  1 .  2
s  = S D P    D  a t  a
c  = I  N    I P  4    2 1  8 .  1 .  1 3  1 .  2
t   = 0    0
m =  a u  d i  o    4 0 0 0    R  T P  / A  V P    8
 - -   b o  u n  d a  r y1
C o  n t  e n  t -  T y  p e  : a p p  l i  c a  t i  o n  / I  S U  P ;  v
e  r s  i o  n =  X -  Q .  7 6  7

^F ^T^D ^@
 -  - b  o u  n d  a r  y1 -  -

  Checking between the sdp part and ISUP part, just after the line m= audio
4000 RTP/AVP 8
we find only one CRLF,and just after this the "- -boundary1"

  Checking in the rfc 2046 i could not clearly find wether this would be
correct, or instead a double CRLF would need to be instead.
  In fact, in one of the examples of rfc 3204 i could see that the same
format is followed and only one CRLF is applied.


Could someone clarify this to me?


  Best regards and thanks,

Carlos





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From exim@www1.ietf.org  Wed Jul  9 03:10:46 2003
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To: "NAKAMURA, Hidefumi" <nakamura.hidefumi@lab.ntt.co.jp>
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Subject: [Sip] Re: [MMUSIC] Appropriate usage of SIP and RTSP
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Hi,

See comments inline:

NAKAMURA, Hidefumi wrote:
> Hi, all.
> 
> I have a question about the relationship between SIP and RTSP. 
> 
> In RFC3261, there is a description that says 
>    "SIP is not a vertically integrated communications system.  SIP is
>    rather a component that can be used with other IETF protocols to
>    build a complete multimedia architecture.  Typically, these
>    architectures will include protocols such as the Real-time Transport
>    Protocol (RTP) (RFC 1889 [28]) for transporting real-time data and
>    providing QoS feedback, the Real-Time streaming protocol (RTSP) (RFC
>    2326 [29]) for controlling delivery of streaming media, ..."
>    (RFC3261 Section 2)
> 
> However, my understanding is that both SIP and RTSP have the capability
> for "discovery of UAs or media streams."
> If I use SIP to establish a stream session and use RTSP to control
> the stream, there will be redundant sequences, i.e. INVITE-OK-ACK and
> SETUP-OK.
> 
> Was there any discussion on how to use these two protocols appropriately?

Not to my knowledge. The SIP and RTSP combination has not been explored, 
however it seem to contain a number of use cases. For example inviting a 
media server into a SIP session or a conference session.

There are significant overlap in SIP and RTSP, mostly in regards to 
session establishment that are done i different ways.

> 
> With SIP application server, network providers can provide various
> services on contents delivery while we will not be able to do so only
> with RTSP, I think.  On the other hand, RTSP has enough capability for
> discovery of media streams and for negotiation of SDPs.
> So, I think, for the simple delivery of contents, RTSP is appropriate to
> use, and for various session control services regarding contents
> delivery, SIP with RTSP is appropriate to use.

What do you mean with "session control services regarding contents 
delivery"?

> 
> Any opinions?
> 
> I wonder why no contents delivery server seems to prepare SIP stack now. 
> Is it because, up to now, there exists only a simple contents delivery
> service?

I believe that no one has looked into this and have had a need.

Best Regards


Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com


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Subject: Re: [Sip] Appropriate usage of SIP and RTSP
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From exim@www1.ietf.org  Wed Jul  9 04:55:57 2003
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From: "Pianigiani, Jacopo (Jacopo)" <jpianigiani@lucent.com>
To: "'Paolo Asprino'" <asprino@coritel.it>,
        "NAKAMURA, Hidefumi"
	 <nakamura.hidefumi@lab.ntt.co.jp>, sip@ietf.org,
        mmusic@ietf.org
Subject: RE: [Sip] Appropriate usage of SIP and RTSP
Date: Wed, 9 Jul 2003 10:53:47 +0200 
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Hello all,
In my undertanding RTSP is suitable for broadcast, unicast or multicast =
streams distribution. The RTSP protocol was ideated for typical content =
delivery, and in effect for this application could be more than =
suitable. But referring to an IMS domain, in my point of view adopting =
RSTP as media streams control, would imply adopting at least a double =
protocol stack in the terminal equipment. The IMS concept was mainly =
driven by the concept - in my understanding - of simplifying the =
terminal (making it simpler from a protocol stack implementation means =
lowering costs and increasing penetration on the market) by adopting a =
single protocol stack in the terminal, and having the RAN and the core =
evolving toward a unified transport mechanism for which an almost =
unified level of control on the network resources allocated to a =
session (whatever session we refer to, voice , multimedia p-t-t call, =
web browsing, multimedia session with voice call and dynamic web =
pages).
So in this sense, my understanding is that the RTSP could be suitable =
for media content, but it would imply again new complexity on the IMS =
UMTS/CDMA terminal itself, which is against the idea of IMS of =
enhancing the services the user can access by having an homogeneous =
protocol stack on the terminal itself for all types of services.

After all, the RTSP , apart from content delivery (of streams), could =
not definitely handle appropriately sessions like Voice calls + =
Interactive Web Pages.=20
Regards

Jacopo Pianigiani
Lucent Technologies - Italy
=E839 335 7350536

-----Original Message-----
From: Paolo Asprino [mailto:asprino@coritel.it]
Sent: Wednesday, July 09, 2003 9:40 AM
To: NAKAMURA, Hidefumi; sip@ietf.org; mmusic@ietf.org
Subject: Re: [Sip] Appropriate usage of SIP and RTSP


Hi all

I'm thinking a solution to join SIP and RTSP in the 3G IMS domain.
In particular in the IMS, I think that to provide multilmedia
service (i.e video on demand), one can estabilish a subscription to the =
service by SIP and then activate the service with RTSP by DIGEST =
autentication with the=20
parameters exchanged early during the subscription (username and =
password)=20

What do you think about that?

Best Regards=20

----- Original Message -----=20
From: "NAKAMURA, Hidefumi" <nakamura.hidefumi@lab.ntt.co.jp>
To: <sip@ietf.org>; <mmusic@ietf.org>
Cc: <hidefumi.nakamura@staff.east.ntt.co.jp>
Sent: Monday, July 07, 2003 9:49 AM
Subject: [Sip] Appropriate usage of SIP and RTSP


> Hi, all.
>=20
> I have a question about the relationship between SIP and RTSP.=20
>=20
> In RFC3261, there is a description that says=20
>    "SIP is not a vertically integrated communications system.  SIP is
>    rather a component that can be used with other IETF protocols to
>    build a complete multimedia architecture.  Typically, these
>    architectures will include protocols such as the Real-time =
Transport
>    Protocol (RTP) (RFC 1889 [28]) for transporting real-time data and
>    providing QoS feedback, the Real-Time streaming protocol (RTSP) =
(RFC
>    2326 [29]) for controlling delivery of streaming media, ..."
>    (RFC3261 Section 2)
>=20
> However, my understanding is that both SIP and RTSP have the =
capability
> for "discovery of UAs or media streams."
> If I use SIP to establish a stream session and use RTSP to control
> the stream, there will be redundant sequences, i.e. INVITE-OK-ACK and
> SETUP-OK.
>=20
> Was there any discussion on how to use these two protocols =
appropriately?
>=20
> With SIP application server, network providers can provide various
> services on contents delivery while we will not be able to do so only
> with RTSP, I think.  On the other hand, RTSP has enough capability =
for
> discovery of media streams and for negotiation of SDPs.
> So, I think, for the simple delivery of contents, RTSP is appropriate =
to
> use, and for various session control services regarding contents
> delivery, SIP with RTSP is appropriate to use.
>=20
> Any opinions?
>=20
> I wonder why no contents delivery server seems to prepare SIP stack =
now.=20
> Is it because, up to now, there exists only a simple contents =
delivery
> service?
>=20
> best regards,
>=20
> --=20
> NAKAMURA, Hidefumi=20
> NTT ServiceIntegration /NetworkServiceSystems Labs.
> tel: +81(422)59 3904; fax:+81(422)60 4012
> ---
>=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


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CC: "NAKAMURA, Hidefumi" <nakamura.hidefumi@lab.ntt.co.jp>, sip@ietf.org,
        mmusic@ietf.org
Subject: Re: [Sip] Appropriate usage of SIP and RTSP
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Paolo Asprino wrote:

> Hi all
> 
> I'm thinking a solution to join SIP and RTSP in the 3G IMS domain.
> In particular in the IMS, I think that to provide multilmedia
> service (i.e video on demand), one can estabilish a subscription to the service by SIP and then activate the service with RTSP by DIGEST autentication with the 
> parameters exchanged early during the subscription (username and password) 
> 
> What do you think about that?

I'm not sure I follow; it seems like you want SIP to be an enrollment 
protocol.

I believe the right path forward is for SIP to setup the session. If 
you want to execute controls, it would be possible to open a control 
session (using an offer/answer exchange), where the control protocol 
is RTSP. In that usage you would only use the PLAY, PAUSE, RECORD, 
GET-PARAMTER and SET-PARAMETER methods. I think the others would not 
be needed.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (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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To: "Pianigiani, Jacopo \(Jacopo\)" <jpianigiani@lucent.com>,
        "NAKAMURA, Hidefumi" <nakamura.hidefumi@lab.ntt.co.jp>, <sip@ietf.org>,
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References: <5160DD6EC1C0D41196D700508B5C167101F805EF@it2020exch001u.it.lucent.com>
Subject: Re: [Sip] Appropriate usage of SIP and RTSP
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From exim@www1.ietf.org  Wed Jul  9 05:55:38 2003
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Date: Wed, 9 Jul 2003 15:00:56 +0530 (IST)
From: Rinchen Tundup <rinchen@cdotd.ernet.in>
To: sip@ietf.org
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Content-Type: TEXT/PLAIN; charset=US-ASCII
Subject: [Sip] Response calculation for Authorization header (fwd)
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hi all ,

I am using the HTTP Authentication RFC code to calculate
"response" for Authorization header. 
I am using estara and SIEMENS proxy to cross check the generated result.

estara and Siemens proxy works fine  ( authentication  works fine )

Message dumps between them :-
SIP/2.0 407 Proxy Authentication Required
Call-ID: 6822d081-8d7d-11d6-996d-0000e23bec27@192.168.6.114
CSeq: 7155859 INVITE
From: <sip:demo@192.168.6.114>;tag=5f1878f0
To: sip:demo@192.168.6.7
Via: SIP/2.0/UDP 192.168.6.114:5060
Content-Length: 0
Proxy-Authenticate:Digest
realm="cdot",nonce="1b8afbde897ce542346164da91a35b",opaque="IWUSipProxy",stale=FALSE,algorithm=MD5,qop="auth"


INVITE sip:demo@192.168.6.7 SIP/2.0
Via: SIP/2.0/UDP 192.168.6.114:5060
From: <sip:demo@192.168.6.114>;tag=5f1878f0
To: sip:demo@192.168.6.7
Contact: sip:demo@192.168.6.114
Call-ID: 6822d081-8d7d-11d6-996d-0000e23bec27@192.168.6.114
CSeq: 7155860 INVITE
Content-Length: 173
Content-Type: application/sdp
User-Agent: eStara SoftPHONE
Supported: com.estara.mux
Proxy-Authorization:  Digest
username="123",realm="cdot",nonce="1b8afbde897ce542346164da91a35b",response="a016c94a34fd3516fe01177eb4dc4a9e",uri="sip:demo@192.168.6.7",opaque="IWUSipProxy"


above works fine 

But using the HTTP auth code :-
   /* Get frm 401 */
	  char * pszNonce = "1b8afbde897ce542346164da91a35b";
	  /* client to choose cnonce */
      char * pszCNonce = "a12";  /* what should be this ?? It can't be
NULL i suppose */
	  /* ph no icw */
      char * pszUser = "123";
      char * pszRealm = "cdot";
      char * pszPass = "123";
      char * pszAlg = "MD5";
      char szNonceCount[9] = "00000001"; /* ?????? what about this */
      char * pszMethod = "INVITE";
      char * pszQop = "auth";
	  /* request uri */
      char * pszURI = "sip:demo@192.168.6.7";
      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);


I get response as ..
Response = 290050605274e103212bd3eac00b10df 

which is not correct ? 

My question is 
1. what should be value of cnonce and nc(nonce count) , since this values
are must for response calulation? 
2.how estara is calculating response without them (as they are not
forming any cnonce or nc parameters)? 
3. how should a client choose cnonce and nc values ?
4. Am i going wrong some where ?  
5. For a given username , passwd , nonce etc how can i check whether my
generated response is corrrect or not ? any freely available tool to check
that  or any standard docs available ?

pls help 
thanks in advance ..


Greetings

Rinchen Tundup      
Delhi




_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From exim@www1.ietf.org  Wed Jul  9 11:14:48 2003
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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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To: "Pianigiani, Jacopo (Jacopo)" <jpianigiani@lucent.com>
CC: sip@ietf.org, mmusic@ietf.org
Subject: Re: [Sip] Appropriate usage of SIP and RTSP
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Hi,

When it comes to IMS and 3G terminals, most will already before having=20
an IMS functionality have support for PSS. PSS is Packet-based Streaming =

Service which is an umbrella kind of standard for streaming service=20
defined in 3GPP targeted at low bit-rates and mobile networks.=20
Specification available from Rel 4 in TS 26.233 and TS 26.234. Already=20
today there exist GSM/GPRS phones that has a PSS client.

So I think that dual stack implementations will not be that impossible=20
to foresee.

However I think the main issue in this discussion is to look at what=20
applications that has need for SIP-RTSP interactions. Based on this we=20
can see if there is solutions easily available or not.

I can agree with Rosenberg that if we where going to design a RTSP=20
functionality protocol today, looking at SIP for session establishment=20
would be interesting. However that opportunity has passed as RTSP has=20
started to be deployed in increasing numbers. I think even the latest=20
Windows media player does support RTSP.


Best Regards

Magnus Westerlund


Pianigiani, Jacopo (Jacopo) wrote:
> Hello all,
> In my undertanding RTSP is suitable for broadcast, unicast or multicast=
 streams distribution. The RTSP protocol was ideated for typical content =
delivery, and in effect for this application could be more than suitable.=
 But referring to an IMS domain, in my point of view adopting RSTP as med=
ia streams control, would imply adopting at least a double protocol stack=
 in the terminal equipment. The IMS concept was mainly driven by the conc=
ept - in my understanding - of simplifying the terminal (making it simple=
r from a protocol stack implementation means lowering costs and increasin=
g penetration on the market) by adopting a single protocol stack in the t=
erminal, and having the RAN and the core evolving toward a unified transp=
ort mechanism for which an almost unified level of control on the network=
 resources allocated to a session (whatever session we refer to, voice , =
multimedia p-t-t call, web browsing, multimedia session with voice call a=
nd dynamic web pages).
> So in this sense, my understanding is that the RTSP could be suitable f=
or media content, but it would imply again new complexity on the IMS UMTS=
/CDMA terminal itself, which is against the idea of IMS of enhancing the =
services the user can access by having an homogeneous protocol stack on t=
he terminal itself for all types of services.
>=20
> After all, the RTSP , apart from content delivery (of streams), could n=
ot definitely handle appropriately sessions like Voice calls + Interactiv=
e Web Pages.=20
> Regards
>=20
> Jacopo Pianigiani
> Lucent Technologies - Italy
> =E839 335 7350536
>=20
> -----Original Message-----
> From: Paolo Asprino [mailto:asprino@coritel.it]
> Sent: Wednesday, July 09, 2003 9:40 AM
> To: NAKAMURA, Hidefumi; sip@ietf.org; mmusic@ietf.org
> Subject: Re: [Sip] Appropriate usage of SIP and RTSP
>=20
>=20
> Hi all
>=20
> I'm thinking a solution to join SIP and RTSP in the 3G IMS domain.
> In particular in the IMS, I think that to provide multilmedia
> service (i.e video on demand), one can estabilish a subscription to the=
 service by SIP and then activate the service with RTSP by DIGEST autenti=
cation with the=20
> parameters exchanged early during the subscription (username and passwo=
rd)=20
>=20
> What do you think about that?
>=20
> Best Regards=20
>=20
> ----- Original Message -----=20
> From: "NAKAMURA, Hidefumi" <nakamura.hidefumi@lab.ntt.co.jp>
> To: <sip@ietf.org>; <mmusic@ietf.org>
> Cc: <hidefumi.nakamura@staff.east.ntt.co.jp>
> Sent: Monday, July 07, 2003 9:49 AM
> Subject: [Sip] Appropriate usage of SIP and RTSP
>=20
>=20
>=20
>>Hi, all.
>>
>>I have a question about the relationship between SIP and RTSP.=20
>>
>>In RFC3261, there is a description that says=20
>>   "SIP is not a vertically integrated communications system.  SIP is
>>   rather a component that can be used with other IETF protocols to
>>   build a complete multimedia architecture.  Typically, these
>>   architectures will include protocols such as the Real-time Transport=

>>   Protocol (RTP) (RFC 1889 [28]) for transporting real-time data and
>>   providing QoS feedback, the Real-Time streaming protocol (RTSP) (RFC=

>>   2326 [29]) for controlling delivery of streaming media, ..."
>>   (RFC3261 Section 2)
>>
>>However, my understanding is that both SIP and RTSP have the capability=

>>for "discovery of UAs or media streams."
>>If I use SIP to establish a stream session and use RTSP to control
>>the stream, there will be redundant sequences, i.e. INVITE-OK-ACK and
>>SETUP-OK.
>>
>>Was there any discussion on how to use these two protocols appropriatel=
y?
>>
>>With SIP application server, network providers can provide various
>>services on contents delivery while we will not be able to do so only
>>with RTSP, I think.  On the other hand, RTSP has enough capability for
>>discovery of media streams and for negotiation of SDPs.
>>So, I think, for the simple delivery of contents, RTSP is appropriate t=
o
>>use, and for various session control services regarding contents
>>delivery, SIP with RTSP is appropriate to use.
>>
>>Any opinions?
>>
>>I wonder why no contents delivery server seems to prepare SIP stack now=
=2E=20
>>Is it because, up to now, there exists only a simple contents delivery
>>service?
>>
>>best regards,
>>
>>--=20
>>NAKAMURA, Hidefumi=20
>>NTT ServiceIntegration /NetworkServiceSystems Labs.
>>tel: +81(422)59 3904; fax:+81(422)60 4012
>>---
>>
>>
>>
>>_______________________________________________
>>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
>=20
>=20
> ---
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)bq=ABb=A2z=1F
>=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


--=20

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.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 exim@www1.ietf.org  Wed Jul  9 12:05:29 2003
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From: Marc Archer <marcher@aastra.com>
To: sip@ietf.org
Subject: RE: [Sip] I-D ACTION:draft-ietf-sip-mib-06.txt
Date: Wed, 9 Jul 2003 11:47:48 -0400 
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Hi,

One more query!

While I was compiling the .MIB file, I got the following Warnings in all the
places where I have Unsigned32. 

--- Warning: unhandled MIB syntax SYNuns32.


I have it in the IMPORTS section of the .MIB file as below:

      IMPORTS      
           MODULE-IDENTITY,      
           OBJECT-TYPE,      
           NOTIFICATION-TYPE,      
           Counter32,      
           Gauge32,      
           TimeTicks,
           Unsigned32,    
           mib-2
                FROM SNMPv2-SMI      

 I have also included in the .INC file 

#condInclude "rfc1902.inc" -- SNMPv2-SMI


It continue further & final stops due to error as it cannot reference any of
those variables.

I am using SMICng version 2.1.03(BOOK)(MS-DOS32), February 12, 1997 &
epilogue 9.1 SNMP stack.

Regards,

Marc Archer

-----Original Message-----
From: Kevin Lingle [mailto:klingle@cisco.com]
Sent: Monday, July 07, 2003 5:10 PM
To: Marc Archer
Cc: sip@ietf.org
Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-mib-06.txt


marc,

we (cisco) used to have essentially the same problem due to
older verison of smicng not recognizing BITS construct.
i believe newer versions of smicng (i think we have v2.2.11 now)
have fixed this afaik.   i know that we used to have to "work around"
this by commenting out BITS syntax and replacing it with OCTET STRING.

i suggest you contact http://www.snmpinfo.com for a definitive answer on
whether your version of smicng is deficient wrt BITS or whether
there is a switch you can turn on to eliminate the compile problem.

btw, i ran the mib modules though smicng before publishing and did
not have any errors reported.... except for the mib numbers: xx, yy
which are eliminated by putting in some arbitrary (high) mib-2 oid
assigned numbers (per dan's earlier suggestion).

kevin

Marc Archer wrote:
> Hi,
> 
> One more query:
> 
> We are trying to integrate the Draft version for SIP MIBs (
> http://www.ietf.org/internet-drafts/draft-ietf-sip-mib-06.txt). In this,
> there is a BITS definition. BITS is not declared in the IMPORTS section
(RFC
> 2578 Section 3.2) of the mib file. I have included RFC 1902 in the .INC
file
> as below:
> 
> #condInclude "rfc1902.inc" -- SNMPv2-SMI
> 
> When I compile, I am getting the error 
> 
> ": f(sip_tc.mib), (3,1) SMI item "BITS" used in SIP-TC, but not defined or
> imported".
> 
> I am using SMICng version 2.1.03(BOOK)(MS-DOS32), February 12, 1997.
> 
> Could someone tell me a fix for this problem? Is there any switch option
> that can take care of this?
> 
> Regards,
> 
> Marc Archer
> 
> -----Original Message-----
> From: Romascanu, Dan (Dan) [mailto:dromasca@avaya.com]
> Sent: Monday, July 07, 2003 9:58 AM
> To: Marc Archer; sip@ietf.org
> Subject: RE: [Sip] I-D ACTION:draft-ietf-sip-mib-06.txt
> 
> 
> Marc,
> 
> 1) The OID assignment for the MIB modules is being done by RFC Editor in
the
> final phases of releasing the RFC. Replace meantime xx with a number of
your
> choice, trying to avoid the already defined numbers, in order to pass
> compilation.
> 2)  RFC 2578 Section 3.2 specifies which symbols must be imported and also
> lists certain pre-defined symbols that must not be imported. The BITS
> construct must not be imported. 
> 
> For more information about standard MIBs review procedures (including
> compilation tools and options) see
>
http://www.ietf.org/internet-drafts/draft-ietf-ops-mib-review-guidelines-01.
> txt.
> 
> Regards,
> 
> Dan
>  
> 
>>-----Original Message-----
>>From: Marc Archer [mailto:marcher@aastra.com]
>>Sent: 07 July, 2003 4:27 PM
>>To: sip@ietf.org
>>Subject: RE: [Sip] I-D ACTION:draft-ietf-sip-mib-06.txt
>>
>>
>>Some comments on the draft
>>
>>TITLE: Has anyone come across a compilation error for not 
>>having a proper
>>IANA assigned number?
>>
>>I am getting error during compilation (Sub-Id for item "sipTC" must be
>>"number" or "name (number)" format) as it is Draft version & 
>>we do not have
>>mib-2 number.  We got to have a number for SIP-COMMON-MIB, SIP-TC &
>>SIP-UA-MIB.
>>
>>           ::= { mib-2 xx } 
>>   -- RFC Ed: replace xx with actual IANA assigned number  
>>
>>Any thoughts? 
>>
>>TITLE: Error due to BITS usage
>>
>>I am experiencing a compilation error due to missing 
>>declaration of BITS in
>>SIP-COMMON-MIB & SIP-TC.
>>
>>There is a BITS usage in the MIB files.
>>
>>              SYNTAX     BITS {      
>>                               other(0),  -- none of the 
>>following     
>>                               udp(1),     
>>                               tcp(2),      
>>                               sctp(3),     
>>                               tls(4)     
>>              }      
>>
>>In the IMPORTS Section, it was not defined; I was getting the 
>>following
>>Error message.
>>
>>SMI item "BITS" used in SIP-TC, but not defined or imported.
>>
>>Since "BITS" is defined in RFC1902.MIB (SNMPv2-SMI), which is already
>>included in the *.INC file, I added "BITS" in the IMPORTS 
>>Section as below:
>>
>>      IMPORTS      
>>           MODULE-IDENTITY,    
>>           mib-2, BITS    
>>                FROM SNMPv2-SMI      
>>
>>The compilation goes much further but it then comes up with 
>>the following
>>error:
>>
>>expected a string, got BITS
>>
>>
>>Any help will be very much appreciated.
>>
>>Regards,
>>
>>Marc Archer
>>
>>-----Original Message-----
>>From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
>>Sent: Wednesday, July 02, 2003 6:57 AM
>>Cc: sip@ietf.org
>>Subject: [Sip] I-D ACTION:draft-ietf-sip-mib-06.txt
>>
>>
>>A New Internet-Draft is available from the on-line Internet-Drafts
>>directories.
>>This draft is a work item of the Session Initiation Protocol 
>>Working Group
>>of the IETF.
>>
>>	Title		: Management Information Base for 
>>Session Initiation
>>
>>                          Protocol
>>	Author(s)	: K. Lingle, J. Maeng, J. Mule, D. Walker
>>	Filename	: draft-ietf-sip-mib-06.txt
>>	Pages		: 104
>>	Date		: 2003-7-1
>>	
>>This memo defines a portion of the Management Information Base (MIB) 
>>for use with network management protocols in the Internet community.  
>>In particular, it describes a set of managed objects that are used 
>>to manage Session Initiation Protocol (SIP) entities, which include 
>>User Agents, Proxy servers, Redirect servers and Registrars.
>>
>>A URL for this Internet-Draft is:
>>http://www.ietf.org/internet-drafts/draft-ietf-sip-mib-06.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-mib-06.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-mib-06.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.
>>
>>_______________________________________________
>>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
> 


-- 
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
  Kevin R. Lingle       919.392.2029
  http://www.klove.com                               http://www.air1.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 exim@www1.ietf.org  Wed Jul  9 12:12:36 2003
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Subject: RE: [Sip] I-D ACTION:draft-ietf-sip-mib-06.txt
Date: Wed, 9 Jul 2003 19:11:01 +0300
Message-ID: <AAB4B3D3CF0F454F98272CBE187FDE2F038A99F6@is0004avexu1.global.avaya.com>
Thread-Topic: [Sip] I-D ACTION:draft-ietf-sip-mib-06.txt
Thread-Index: AcNGM9W0NbY78HczTZ+46o3yr3GUiQAACDZQ
From: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
To: "Marc Archer" <marcher@aastra.com>, <sip@ietf.org>
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See section 4.6.1.1 in =
http://www.ietf.org/internet-drafts/draft-ietf-ops-mib-review-guidelines-=
01.txt for the usage of Unsigned32, and Appendix C for the use of SMIng =
for MIB modules compilation.

Regards,

Dan


> -----Original Message-----
> From: Marc Archer [mailto:marcher@aastra.com]
> Sent: 09 July, 2003 6:48 PM
> To: sip@ietf.org
> Subject: RE: [Sip] I-D ACTION:draft-ietf-sip-mib-06.txt
>=20
>=20
> Hi,
>=20
> One more query!
>=20
> While I was compiling the .MIB file, I got the following=20
> Warnings in all the
> places where I have Unsigned32.=20
>=20
> --- Warning: unhandled MIB syntax SYNuns32.
>=20
>=20
> I have it in the IMPORTS section of the .MIB file as below:
>=20
>       IMPORTS     =20
>            MODULE-IDENTITY,     =20
>            OBJECT-TYPE,     =20
>            NOTIFICATION-TYPE,     =20
>            Counter32,     =20
>            Gauge32,     =20
>            TimeTicks,
>            Unsigned32,   =20
>            mib-2
>                 FROM SNMPv2-SMI     =20
>=20
>  I have also included in the .INC file=20
>=20
> #condInclude "rfc1902.inc" -- SNMPv2-SMI
>=20
>=20
> It continue further & final stops due to error as it cannot=20
> reference any of
> those variables.
>=20
> I am using SMICng version 2.1.03(BOOK)(MS-DOS32), February 12, 1997 &
> epilogue 9.1 SNMP stack.
>=20
> Regards,
>=20
> Marc Archer
>=20
> -----Original Message-----
> From: Kevin Lingle [mailto:klingle@cisco.com]
> Sent: Monday, July 07, 2003 5:10 PM
> To: Marc Archer
> Cc: sip@ietf.org
> Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-mib-06.txt
>=20
>=20
> marc,
>=20
> we (cisco) used to have essentially the same problem due to
> older verison of smicng not recognizing BITS construct.
> i believe newer versions of smicng (i think we have v2.2.11 now)
> have fixed this afaik.   i know that we used to have to "work around"
> this by commenting out BITS syntax and replacing it with OCTET STRING.
>=20
> i suggest you contact http://www.snmpinfo.com for a=20
> definitive answer on
> whether your version of smicng is deficient wrt BITS or whether
> there is a switch you can turn on to eliminate the compile problem.
>=20
> btw, i ran the mib modules though smicng before publishing and did
> not have any errors reported.... except for the mib numbers: xx, yy
> which are eliminated by putting in some arbitrary (high) mib-2 oid
> assigned numbers (per dan's earlier suggestion).
>=20
> kevin
>=20
> Marc Archer wrote:
> > Hi,
> >=20
> > One more query:
> >=20
> > We are trying to integrate the Draft version for SIP MIBs (
> >=20
> http://www.ietf.org/internet-drafts/draft-ietf-sip-mib-06.txt)
> . In this,
> > there is a BITS definition. BITS is not declared in the=20
> IMPORTS section
> (RFC
> > 2578 Section 3.2) of the mib file. I have included RFC 1902=20
> in the .INC
> file
> > as below:
> >=20
> > #condInclude "rfc1902.inc" -- SNMPv2-SMI
> >=20
> > When I compile, I am getting the error=20
> >=20
> > ": f(sip_tc.mib), (3,1) SMI item "BITS" used in SIP-TC, but=20
> not defined or
> > imported".
> >=20
> > I am using SMICng version 2.1.03(BOOK)(MS-DOS32), February 12, 1997.
> >=20
> > Could someone tell me a fix for this problem? Is there any=20
> switch option
> > that can take care of this?
> >=20
> > Regards,
> >=20
> > Marc Archer
> >=20
> > -----Original Message-----
> > From: Romascanu, Dan (Dan) [mailto:dromasca@avaya.com]
> > Sent: Monday, July 07, 2003 9:58 AM
> > To: Marc Archer; sip@ietf.org
> > Subject: RE: [Sip] I-D ACTION:draft-ietf-sip-mib-06.txt
> >=20
> >=20
> > Marc,
> >=20
> > 1) The OID assignment for the MIB modules is being done by=20
> RFC Editor in
> the
> > final phases of releasing the RFC. Replace meantime xx with=20
> a number of
> your
> > choice, trying to avoid the already defined numbers, in=20
> order to pass
> > compilation.
> > 2)  RFC 2578 Section 3.2 specifies which symbols must be=20
> imported and also
> > lists certain pre-defined symbols that must not be=20
> imported. The BITS
> > construct must not be imported.=20
> >=20
> > For more information about standard MIBs review procedures=20
> (including
> > compilation tools and options) see
> >
> http://www.ietf.org/internet-drafts/draft-ietf-ops-mib-review-
> guidelines-01.
> > txt.
> >=20
> > Regards,
> >=20
> > Dan
> > =20
> >=20
> >>-----Original Message-----
> >>From: Marc Archer [mailto:marcher@aastra.com]
> >>Sent: 07 July, 2003 4:27 PM
> >>To: sip@ietf.org
> >>Subject: RE: [Sip] I-D ACTION:draft-ietf-sip-mib-06.txt
> >>
> >>
> >>Some comments on the draft
> >>
> >>TITLE: Has anyone come across a compilation error for not=20
> >>having a proper
> >>IANA assigned number?
> >>
> >>I am getting error during compilation (Sub-Id for item=20
> "sipTC" must be
> >>"number" or "name (number)" format) as it is Draft version &=20
> >>we do not have
> >>mib-2 number.  We got to have a number for SIP-COMMON-MIB, SIP-TC &
> >>SIP-UA-MIB.
> >>
> >>           ::=3D { mib-2 xx }=20
> >>   -- RFC Ed: replace xx with actual IANA assigned number =20
> >>
> >>Any thoughts?=20
> >>
> >>TITLE: Error due to BITS usage
> >>
> >>I am experiencing a compilation error due to missing=20
> >>declaration of BITS in
> >>SIP-COMMON-MIB & SIP-TC.
> >>
> >>There is a BITS usage in the MIB files.
> >>
> >>              SYNTAX     BITS {     =20
> >>                               other(0),  -- none of the=20
> >>following    =20
> >>                               udp(1),    =20
> >>                               tcp(2),     =20
> >>                               sctp(3),    =20
> >>                               tls(4)    =20
> >>              }     =20
> >>
> >>In the IMPORTS Section, it was not defined; I was getting the=20
> >>following
> >>Error message.
> >>
> >>SMI item "BITS" used in SIP-TC, but not defined or imported.
> >>
> >>Since "BITS" is defined in RFC1902.MIB (SNMPv2-SMI), which=20
> is already
> >>included in the *.INC file, I added "BITS" in the IMPORTS=20
> >>Section as below:
> >>
> >>      IMPORTS     =20
> >>           MODULE-IDENTITY,   =20
> >>           mib-2, BITS   =20
> >>                FROM SNMPv2-SMI     =20
> >>
> >>The compilation goes much further but it then comes up with=20
> >>the following
> >>error:
> >>
> >>expected a string, got BITS
> >>
> >>
> >>Any help will be very much appreciated.
> >>
> >>Regards,
> >>
> >>Marc Archer
> >>
> >>-----Original Message-----
> >>From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
> >>Sent: Wednesday, July 02, 2003 6:57 AM
> >>Cc: sip@ietf.org
> >>Subject: [Sip] I-D ACTION:draft-ietf-sip-mib-06.txt
> >>
> >>
> >>A New Internet-Draft is available from the on-line Internet-Drafts
> >>directories.
> >>This draft is a work item of the Session Initiation Protocol=20
> >>Working Group
> >>of the IETF.
> >>
> >>	Title		: Management Information Base for=20
> >>Session Initiation
> >>
> >>                          Protocol
> >>	Author(s)	: K. Lingle, J. Maeng, J. Mule, D. Walker
> >>	Filename	: draft-ietf-sip-mib-06.txt
> >>	Pages		: 104
> >>	Date		: 2003-7-1
> >>=09
> >>This memo defines a portion of the Management Information=20
> Base (MIB)=20
> >>for use with network management protocols in the Internet=20
> community. =20
> >>In particular, it describes a set of managed objects that are used=20
> >>to manage Session Initiation Protocol (SIP) entities, which include=20
> >>User Agents, Proxy servers, Redirect servers and Registrars.
> >>
> >>A URL for this Internet-Draft is:
> >>http://www.ietf.org/internet-drafts/draft-ietf-sip-mib-06.txt
> >>
> >>To remove yourself from the IETF Announcement list, send a=20
> message to=20
> >>ietf-announce-request with the word unsubscribe in the body=20
> >>of the message.
> >>
> >>Internet-Drafts are also available by anonymous FTP. Login=20
> >>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-mib-06.txt".
> >>
> >>A list of Internet-Drafts directories can be found in
> >>http://www.ietf.org/shadow.html=20
> >>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-mib-06.txt".
> >>=09
> >>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=20
> >>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.
> >>	=09
> >>	=09
> >>Below is the data which will enable a MIME compliant mail reader
> >>implementation to automatically retrieve the ASCII version of the
> >>Internet-Draft.
> >>
> >>_______________________________________________
> >>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
> >=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
>=20
>=20
> --=20
> =
=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D=
-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-
> =3D-=3D-=3D-=3D-=3D
>   Kevin R. Lingle       919.392.2029
>   http://www.klove.com                              =20
http://www.air1.com
=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D=
-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D-=3D

_______________________________________________
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: Kevin Lingle <klingle@cisco.com>
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CC: sip@ietf.org
Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-mib-06.txt
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marc,

i'm sorry for the trouble you are having, but i fail to see
the problem with the mib module.   Unsigned32 is defined
in SNMPv2-SMI so the import is proper.

my understanding is that SMICng inherently understands
the "standard" constructs from the SMI - like Unsigned32.

i don't really know what to tell you.  i've never seen any
such compiler warnings with any revision of these or other
mibs where Unsigned32 syntax exists - even with our older
version of SMIC.

kevin

Marc Archer wrote:
> Hi,
> 
> One more query!
> 
> While I was compiling the .MIB file, I got the following Warnings in all the
> places where I have Unsigned32. 
> 
> --- Warning: unhandled MIB syntax SYNuns32.
> 
> 
> I have it in the IMPORTS section of the .MIB file as below:
> 
>       IMPORTS      
>            MODULE-IDENTITY,      
>            OBJECT-TYPE,      
>            NOTIFICATION-TYPE,      
>            Counter32,      
>            Gauge32,      
>            TimeTicks,
>            Unsigned32,    
>            mib-2
>                 FROM SNMPv2-SMI      
> 
>  I have also included in the .INC file 
> 
> #condInclude "rfc1902.inc" -- SNMPv2-SMI
> 
> 
> It continue further & final stops due to error as it cannot reference any of
> those variables.
> 
> I am using SMICng version 2.1.03(BOOK)(MS-DOS32), February 12, 1997 &
> epilogue 9.1 SNMP stack.
> 
> Regards,
> 
> Marc Archer
> 
> -----Original Message-----
> From: Kevin Lingle [mailto:klingle@cisco.com]
> Sent: Monday, July 07, 2003 5:10 PM
> To: Marc Archer
> Cc: sip@ietf.org
> Subject: Re: [Sip] I-D ACTION:draft-ietf-sip-mib-06.txt
> 
> 
> marc,
> 
> we (cisco) used to have essentially the same problem due to
> older verison of smicng not recognizing BITS construct.
> i believe newer versions of smicng (i think we have v2.2.11 now)
> have fixed this afaik.   i know that we used to have to "work around"
> this by commenting out BITS syntax and replacing it with OCTET STRING.
> 
> i suggest you contact http://www.snmpinfo.com for a definitive answer on
> whether your version of smicng is deficient wrt BITS or whether
> there is a switch you can turn on to eliminate the compile problem.
> 
> btw, i ran the mib modules though smicng before publishing and did
> not have any errors reported.... except for the mib numbers: xx, yy
> which are eliminated by putting in some arbitrary (high) mib-2 oid
> assigned numbers (per dan's earlier suggestion).
> 
> kevin
> 
> Marc Archer wrote:
> 
>>Hi,
>>
>>One more query:
>>
>>We are trying to integrate the Draft version for SIP MIBs (
>>http://www.ietf.org/internet-drafts/draft-ietf-sip-mib-06.txt). In this,
>>there is a BITS definition. BITS is not declared in the IMPORTS section
> 
> (RFC
> 
>>2578 Section 3.2) of the mib file. I have included RFC 1902 in the .INC
> 
> file
> 
>>as below:
>>
>>#condInclude "rfc1902.inc" -- SNMPv2-SMI
>>
>>When I compile, I am getting the error 
>>
>>": f(sip_tc.mib), (3,1) SMI item "BITS" used in SIP-TC, but not defined or
>>imported".
>>
>>I am using SMICng version 2.1.03(BOOK)(MS-DOS32), February 12, 1997.
>>
>>Could someone tell me a fix for this problem? Is there any switch option
>>that can take care of this?
>>
>>Regards,
>>
>>Marc Archer
>>
>>-----Original Message-----
>>From: Romascanu, Dan (Dan) [mailto:dromasca@avaya.com]
>>Sent: Monday, July 07, 2003 9:58 AM
>>To: Marc Archer; sip@ietf.org
>>Subject: RE: [Sip] I-D ACTION:draft-ietf-sip-mib-06.txt
>>
>>
>>Marc,
>>
>>1) The OID assignment for the MIB modules is being done by RFC Editor in
> 
> the
> 
>>final phases of releasing the RFC. Replace meantime xx with a number of
> 
> your
> 
>>choice, trying to avoid the already defined numbers, in order to pass
>>compilation.
>>2)  RFC 2578 Section 3.2 specifies which symbols must be imported and also
>>lists certain pre-defined symbols that must not be imported. The BITS
>>construct must not be imported. 
>>
>>For more information about standard MIBs review procedures (including
>>compilation tools and options) see
>>
> 
> http://www.ietf.org/internet-drafts/draft-ietf-ops-mib-review-guidelines-01.
> 
>>txt.
>>
>>Regards,
>>
>>Dan
>> 
>>
>>
>>>-----Original Message-----
>>>From: Marc Archer [mailto:marcher@aastra.com]
>>>Sent: 07 July, 2003 4:27 PM
>>>To: sip@ietf.org
>>>Subject: RE: [Sip] I-D ACTION:draft-ietf-sip-mib-06.txt
>>>
>>>
>>>Some comments on the draft
>>>
>>>TITLE: Has anyone come across a compilation error for not 
>>>having a proper
>>>IANA assigned number?
>>>
>>>I am getting error during compilation (Sub-Id for item "sipTC" must be
>>>"number" or "name (number)" format) as it is Draft version & 
>>>we do not have
>>>mib-2 number.  We got to have a number for SIP-COMMON-MIB, SIP-TC &
>>>SIP-UA-MIB.
>>>
>>>          ::= { mib-2 xx } 
>>>  -- RFC Ed: replace xx with actual IANA assigned number  
>>>
>>>Any thoughts? 
>>>
>>>TITLE: Error due to BITS usage
>>>
>>>I am experiencing a compilation error due to missing 
>>>declaration of BITS in
>>>SIP-COMMON-MIB & SIP-TC.
>>>
>>>There is a BITS usage in the MIB files.
>>>
>>>             SYNTAX     BITS {      
>>>                              other(0),  -- none of the 
>>>following     
>>>                              udp(1),     
>>>                              tcp(2),      
>>>                              sctp(3),     
>>>                              tls(4)     
>>>             }      
>>>
>>>In the IMPORTS Section, it was not defined; I was getting the 
>>>following
>>>Error message.
>>>
>>>SMI item "BITS" used in SIP-TC, but not defined or imported.
>>>
>>>Since "BITS" is defined in RFC1902.MIB (SNMPv2-SMI), which is already
>>>included in the *.INC file, I added "BITS" in the IMPORTS 
>>>Section as below:
>>>
>>>     IMPORTS      
>>>          MODULE-IDENTITY,    
>>>          mib-2, BITS    
>>>               FROM SNMPv2-SMI      
>>>
>>>The compilation goes much further but it then comes up with 
>>>the following
>>>error:
>>>
>>>expected a string, got BITS
>>>
>>>
>>>Any help will be very much appreciated.
>>>
>>>Regards,
>>>
>>>Marc Archer
>>>
>>>-----Original Message-----
>>>From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
>>>Sent: Wednesday, July 02, 2003 6:57 AM
>>>Cc: sip@ietf.org
>>>Subject: [Sip] I-D ACTION:draft-ietf-sip-mib-06.txt
>>>
>>>
>>>A New Internet-Draft is available from the on-line Internet-Drafts
>>>directories.
>>>This draft is a work item of the Session Initiation Protocol 
>>>Working Group
>>>of the IETF.
>>>
>>>	Title		: Management Information Base for 
>>>Session Initiation
>>>
>>>                         Protocol
>>>	Author(s)	: K. Lingle, J. Maeng, J. Mule, D. Walker
>>>	Filename	: draft-ietf-sip-mib-06.txt
>>>	Pages		: 104
>>>	Date		: 2003-7-1
>>>	
>>>This memo defines a portion of the Management Information Base (MIB) 
>>>for use with network management protocols in the Internet community.  
>>>In particular, it describes a set of managed objects that are used 
>>>to manage Session Initiation Protocol (SIP) entities, which include 
>>>User Agents, Proxy servers, Redirect servers and Registrars.
>>>
>>>A URL for this Internet-Draft is:
>>>http://www.ietf.org/internet-drafts/draft-ietf-sip-mib-06.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-mib-06.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-mib-06.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.
>>>
>>>_______________________________________________
>>>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
>>
> 
> 
> 


-- 
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
  Kevin R. Lingle       919.392.2029
  http://www.klove.com                               http://www.air1.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 exim@www1.ietf.org  Wed Jul  9 14:43:00 2003
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Yutaka Takeda'" <takeday@kmerl.com>, sip@ietf.org, mmusic@ietf.org
Subject: RE: [MMUSIC] Re: [Sip] Appropriate usage of SIP and RTSP
Date: Wed, 9 Jul 2003 14:40:50 -0400 
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I've been thinking for a long time about what an appropriate
mechanism to provide a voice/videomail client on a sipphone
would be.  It's really clear that you record and play messages
with VCR like functionality.  For that, you need start/stop/pause/
rewind/fast forward, and so I was thinking RTSP.  On the other
hand, you surely establish the session to the voicemail server
with a sip call.

Thus, I like the way Jonathan is thinking, but you also need
message selection/creation/deletion functions, and for those
I was thinking KPML, and then you wonder if you just use that
as the signalling mechanism for the record/play, and I get
confused.

Brian

> -----Original Message-----
> From: Yutaka Takeda [mailto:takeday@kmerl.com]
> Sent: Wednesday, July 09, 2003 2:11 PM
> To: sip@ietf.org; mmusic@ietf.org
> Subject: RE: [MMUSIC] Re: [Sip] Appropriate usage of SIP and RTSP
>=20
>=20
> The applications that have need for SIP and RTSP features I
> can imagine are:
>=20
> - Voice&video mail (similar to 'sipum' but VM server is in=20
> the SIP phone)
> - Personal broadcast station from my PC at home.
> - Watching video recorded with VCR at home or office from remote...
>=20
> What I am not sure is if RTSP has any solution or a framework for a
> RTSP server behind a NAT. If not, the solution is likely to=20
> be a repeat
> of what SIP provides. I thought TURN could be used in order=20
> for signaling
> message to reach RTSP server behind a NAT, but I noticed that=20
> it is not=20
> applicable (purposefully) for a server running behind a NAT because of
> a security consideration. Lots of issues for RTSP in this area alone?
>=20
> I believe a lot of people are interested in this discussion.
>=20
> Best regards,
> Yutaka
>=20
>=20
> > -----Original Message-----
> > From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> > Sent: Wednesday, July 09, 2003 5:53 AM
> > To: Pianigiani, Jacopo (Jacopo)
> > Cc: sip@ietf.org; mmusic@ietf.org
> > Subject: [MMUSIC] Re: [Sip] Appropriate usage of SIP and RTSP
> >=20
> >=20
> > Hi,
> >=20
> > When it comes to IMS and 3G terminals, most will already=20
> > before having=20
> > an IMS functionality have support for PSS. PSS is=20
> > Packet-based Streaming=20
> > Service which is an umbrella kind of standard for streaming service=20
> > defined in 3GPP targeted at low bit-rates and mobile networks.=20
> > Specification available from Rel 4 in TS 26.233 and TS=20
> > 26.234. Already=20
> > today there exist GSM/GPRS phones that has a PSS client.
> >=20
> > So I think that dual stack implementations will not be that=20
> > impossible=20
> > to foresee.
> >=20
> > However I think the main issue in this discussion is to=20
> look at what=20
> > applications that has need for SIP-RTSP interactions. Based=20
> > on this we=20
> > can see if there is solutions easily available or not.
> >=20
> > I can agree with Rosenberg that if we where going to design a RTSP=20
> > functionality protocol today, looking at SIP for session=20
> > establishment=20
> > would be interesting. However that opportunity has passed=20
> as RTSP has=20
> > started to be deployed in increasing numbers. I think even=20
> the latest=20
> > Windows media player does support RTSP.
> >=20
> >=20
> > Best Regards
> >=20
> > Magnus Westerlund
> >=20
> >=20
> > Pianigiani, Jacopo (Jacopo) wrote:
> > > Hello all,
> > > In my undertanding RTSP is suitable for broadcast, unicast=20
> > or multicast streams distribution. The RTSP protocol was=20
> > ideated for typical content delivery, and in effect for this=20
> > application could be more than suitable. But referring to an=20
> > IMS domain, in my point of view adopting RSTP as media=20
> > streams control, would imply adopting at least a double=20
> > protocol stack in the terminal equipment. The IMS concept was=20
> > mainly driven by the concept - in my understanding - of=20
> > simplifying the terminal (making it simpler from a protocol=20
> > stack implementation means lowering costs and increasing=20
> > penetration on the market) by adopting a single protocol=20
> > stack in the terminal, and having the RAN and the core=20
> > evolving toward a unified transport mechanism for which an=20
> > almost unified level of control on the network resources=20
> > allocated to a session (whatever session we refer to, voice ,=20
> > multimedia p-t-t call, web browsing, multimedia session with=20
> > voice call and dynamic web pages).
> > > So in this sense, my understanding is that the RTSP could=20
> > be suitable for media content, but it would imply again new=20
> > complexity on the IMS UMTS/CDMA terminal itself, which is=20
> > against the idea of IMS of enhancing the services the user=20
> > can access by having an homogeneous protocol stack on the=20
> > terminal itself for all types of services.
> > >=20
> > > After all, the RTSP , apart from content delivery (of=20
> > streams), could not definitely handle appropriately sessions=20
> > like Voice calls + Interactive Web Pages.=20
> > > Regards
> > >=20
> > > Jacopo Pianigiani
> > > Lucent Technologies - Italy
> > > =E839 335 7350536
> > >=20
> > > -----Original Message-----
> > > From: Paolo Asprino [mailto:asprino@coritel.it]
> > > Sent: Wednesday, July 09, 2003 9:40 AM
> > > To: NAKAMURA, Hidefumi; sip@ietf.org; mmusic@ietf.org
> > > Subject: Re: [Sip] Appropriate usage of SIP and RTSP
> > >=20
> > >=20
> > > Hi all
> > >=20
> > > I'm thinking a solution to join SIP and RTSP in the 3G IMS domain.
> > > In particular in the IMS, I think that to provide multilmedia
> > > service (i.e video on demand), one can estabilish a=20
> > subscription to the service by SIP and then activate the=20
> > service with RTSP by DIGEST autentication with the=20
> > > parameters exchanged early during the subscription=20
> > (username and password)=20
> > >=20
> > > What do you think about that?
> > >=20
> > > Best Regards=20
> > >=20
> > > ----- Original Message -----=20
> > > From: "NAKAMURA, Hidefumi" <nakamura.hidefumi@lab.ntt.co.jp>
> > > To: <sip@ietf.org>; <mmusic@ietf.org>
> > > Cc: <hidefumi.nakamura@staff.east.ntt.co.jp>
> > > Sent: Monday, July 07, 2003 9:49 AM
> > > Subject: [Sip] Appropriate usage of SIP and RTSP
> > >=20
> > >=20
> > >=20
> > >>Hi, all.
> > >>
> > >>I have a question about the relationship between SIP and RTSP.=20
> > >>
> > >>In RFC3261, there is a description that says=20
> > >>   "SIP is not a vertically integrated communications=20
> > system.  SIP is
> > >>   rather a component that can be used with other IETF=20
> protocols to
> > >>   build a complete multimedia architecture.  Typically, these
> > >>   architectures will include protocols such as the=20
> > Real-time Transport
> > >>   Protocol (RTP) (RFC 1889 [28]) for transporting=20
> > real-time data and
> > >>   providing QoS feedback, the Real-Time streaming protocol=20
> > (RTSP) (RFC
> > >>   2326 [29]) for controlling delivery of streaming media, ..."
> > >>   (RFC3261 Section 2)
> > >>
> > >>However, my understanding is that both SIP and RTSP have=20
> > the capability
> > >>for "discovery of UAs or media streams."
> > >>If I use SIP to establish a stream session and use RTSP to control
> > >>the stream, there will be redundant sequences, i.e.=20
> > INVITE-OK-ACK and
> > >>SETUP-OK.
> > >>
> > >>Was there any discussion on how to use these two protocols=20
> > appropriately?
> > >>
> > >>With SIP application server, network providers can provide various
> > >>services on contents delivery while we will not be able to=20
> > do so only
> > >>with RTSP, I think.  On the other hand, RTSP has enough=20
> > capability for
> > >>discovery of media streams and for negotiation of SDPs.
> > >>So, I think, for the simple delivery of contents, RTSP is=20
> > appropriate to
> > >>use, and for various session control services regarding contents
> > >>delivery, SIP with RTSP is appropriate to use.
> > >>
> > >>Any opinions?
> > >>
> > >>I wonder why no contents delivery server seems to prepare=20
> > SIP stack now.=20
> > >>Is it because, up to now, there exists only a simple=20
> > contents delivery
> > >>service?
> > >>
> > >>best regards,
> > >>
> > >>--=20
> > >>NAKAMURA, Hidefumi=20
> > >>NTT ServiceIntegration /NetworkServiceSystems Labs.
> > >>tel: +81(422)59 3904; fax:+81(422)60 4012
> > >>---
> > >>
> > >>
> > >>
> > >>_______________________________________________
> > >>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=20
> application of sip
> > >>
> > >=20
> > >=20
> > >=20
> > > ---
> > > Outgoing mail is certified Virus Free.
> > > Checked by AVG anti-virus system (http://www.grisoft.com).
> > > Version: 6.0.495 / Virus Database: 294 - Release Date:=20
> > 30/06/2003J*fj)bz	b=B2=D8m=B6>?=FF=0C> > 0=D6'=AD~S=E0=FEf=A2-f=A7=FE=
X=AC=B6)=DF=A3=FB"=A58b=B2X=AC=B6+=1F=A2=B3DY=D7=AF
> > zZ)(tm)=E9=ED=A1=FBay=CA+y"=0F>=BA-=A1=CA%R=C7=ACS~=A6=A6W=A6z{h=AE=C7=
,r?n(tm)=B8sy=DBY=A2=BA=AEz=CBb=A2{(=9D=CB=AB
> > =AD=E9=ED=B2*T=B1=EB"=A6~=A7''=AD~S=E0~S=E7{=07^=BD=E9h=A6g=A7=B6=CA'=
=B6=17s=A6(tm)bq=ABb=A2z=1F
> > >=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=20
> application of sip
> > >=20
> >=20
> >=20
> > --=20
> >=20
> > Magnus Westerlund
> >=20
> > Multimedia Technologies, Ericsson Research EAB/TVA/A
> >=20
> ----------------------------------------------------------------------
> > Ericsson AB                | Phone +46 8 4048287
> > Torshamsgatan 23           | Fax   +46 8 7575550
> > S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> >=20
> >=20
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www1.ietf.org/mailman/listinfo/mmusic
> >=20
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
>=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



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Date: Wed, 9 Jul 2003 11:53:39 -0700 (PDT)
To: "Rosen, Brian" <Brian.Rosen@marconi.com>
Cc: "'Yutaka Takeda'" <takeday@kmerl.com>, sip@ietf.org, mmusic@ietf.org
Subject: RE: [MMUSIC] Re: [Sip] Appropriate usage of SIP and RTSP
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Hmm. I'd always thought of SIP as a rendezvous
protocol so this seems perfectly natural to me.
Is the general problem here that the SDP
announcement for an RTSP controlled session isn't
defined? Or is the issue that people would like to
piggyback both the RTSP and RTP session
information in the SIP SDP? Or both?=20

Of course if the announcement for RTSP is not
specified, it sort of begs the question of why
you'd want to just define RTSP semantics... why
not HTTP semantics, etc too? (in which case, it
seems to me that an over-loaded SIP SDP
announcment is probably the wrong thing to do).

=09       Mike

Rosen, Brian writes:
 > I've been thinking for a long time about what an appropriate
 > mechanism to provide a voice/videomail client on a sipphone
 > would be.  It's really clear that you record and play messages
 > with VCR like functionality.  For that, you need start/stop/pause/
 > rewind/fast forward, and so I was thinking RTSP.  On the other
 > hand, you surely establish the session to the voicemail server
 > with a sip call.
 >=20
 > Thus, I like the way Jonathan is thinking, but you also need
 > message selection/creation/deletion functions, and for those
 > I was thinking KPML, and then you wonder if you just use that
 > as the signalling mechanism for the record/play, and I get
 > confused.
 >=20
 > Brian
 >=20
 > > -----Original Message-----
 > > From: Yutaka Takeda [mailto:takeday@kmerl.com]
 > > Sent: Wednesday, July 09, 2003 2:11 PM
 > > To: sip@ietf.org; mmusic@ietf.org
 > > Subject: RE: [MMUSIC] Re: [Sip] Appropriate usage of SIP and RTSP
 > >=20
 > >=20
 > > The applications that have need for SIP and RTSP features I
 > > can imagine are:
 > >=20
 > > - Voice&video mail (similar to 'sipum' but VM server is in=20
 > > the SIP phone)
 > > - Personal broadcast station from my PC at home.
 > > - Watching video recorded with VCR at home or office from remote..=
.
 > >=20
 > > What I am not sure is if RTSP has any solution or a framework for =
a
 > > RTSP server behind a NAT. If not, the solution is likely to=20
 > > be a repeat
 > > of what SIP provides. I thought TURN could be used in order=20
 > > for signaling
 > > message to reach RTSP server behind a NAT, but I noticed that=20
 > > it is not=20
 > > applicable (purposefully) for a server running behind a NAT becaus=
e of
 > > a security consideration. Lots of issues for RTSP in this area alo=
ne?
 > >=20
 > > I believe a lot of people are interested in this discussion.
 > >=20
 > > Best regards,
 > > Yutaka
 > >=20
 > >=20
 > > > -----Original Message-----
 > > > From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
 > > > Sent: Wednesday, July 09, 2003 5:53 AM
 > > > To: Pianigiani, Jacopo (Jacopo)
 > > > Cc: sip@ietf.org; mmusic@ietf.org
 > > > Subject: [MMUSIC] Re: [Sip] Appropriate usage of SIP and RTSP
 > > >=20
 > > >=20
 > > > Hi,
 > > >=20
 > > > When it comes to IMS and 3G terminals, most will already=20
 > > > before having=20
 > > > an IMS functionality have support for PSS. PSS is=20
 > > > Packet-based Streaming=20
 > > > Service which is an umbrella kind of standard for streaming serv=
ice=20
 > > > defined in 3GPP targeted at low bit-rates and mobile networks.=20=

 > > > Specification available from Rel 4 in TS 26.233 and TS=20
 > > > 26.234. Already=20
 > > > today there exist GSM/GPRS phones that has a PSS client.
 > > >=20
 > > > So I think that dual stack implementations will not be that=20
 > > > impossible=20
 > > > to foresee.
 > > >=20
 > > > However I think the main issue in this discussion is to=20
 > > look at what=20
 > > > applications that has need for SIP-RTSP interactions. Based=20
 > > > on this we=20
 > > > can see if there is solutions easily available or not.
 > > >=20
 > > > I can agree with Rosenberg that if we where going to design a RT=
SP=20
 > > > functionality protocol today, looking at SIP for session=20
 > > > establishment=20
 > > > would be interesting. However that opportunity has passed=20
 > > as RTSP has=20
 > > > started to be deployed in increasing numbers. I think even=20
 > > the latest=20
 > > > Windows media player does support RTSP.
 > > >=20
 > > >=20
 > > > Best Regards
 > > >=20
 > > > Magnus Westerlund
 > > >=20
 > > >=20
 > > > Pianigiani, Jacopo (Jacopo) wrote:
 > > > > Hello all,
 > > > > In my undertanding RTSP is suitable for broadcast, unicast=20
 > > > or multicast streams distribution. The RTSP protocol was=20
 > > > ideated for typical content delivery, and in effect for this=20
 > > > application could be more than suitable. But referring to an=20
 > > > IMS domain, in my point of view adopting RSTP as media=20
 > > > streams control, would imply adopting at least a double=20
 > > > protocol stack in the terminal equipment. The IMS concept was=20=

 > > > mainly driven by the concept - in my understanding - of=20
 > > > simplifying the terminal (making it simpler from a protocol=20
 > > > stack implementation means lowering costs and increasing=20
 > > > penetration on the market) by adopting a single protocol=20
 > > > stack in the terminal, and having the RAN and the core=20
 > > > evolving toward a unified transport mechanism for which an=20
 > > > almost unified level of control on the network resources=20
 > > > allocated to a session (whatever session we refer to, voice ,=20=

 > > > multimedia p-t-t call, web browsing, multimedia session with=20
 > > > voice call and dynamic web pages).
 > > > > So in this sense, my understanding is that the RTSP could=20
 > > > be suitable for media content, but it would imply again new=20
 > > > complexity on the IMS UMTS/CDMA terminal itself, which is=20
 > > > against the idea of IMS of enhancing the services the user=20
 > > > can access by having an homogeneous protocol stack on the=20
 > > > terminal itself for all types of services.
 > > > >=20
 > > > > After all, the RTSP , apart from content delivery (of=20
 > > > streams), could not definitely handle appropriately sessions=20
 > > > like Voice calls + Interactive Web Pages.=20
 > > > > Regards
 > > > >=20
 > > > > Jacopo Pianigiani
 > > > > Lucent Technologies - Italy
 > > > > =E839 335 7350536
 > > > >=20
 > > > > -----Original Message-----
 > > > > From: Paolo Asprino [mailto:asprino@coritel.it]
 > > > > Sent: Wednesday, July 09, 2003 9:40 AM
 > > > > To: NAKAMURA, Hidefumi; sip@ietf.org; mmusic@ietf.org
 > > > > Subject: Re: [Sip] Appropriate usage of SIP and RTSP
 > > > >=20
 > > > >=20
 > > > > Hi all
 > > > >=20
 > > > > I'm thinking a solution to join SIP and RTSP in the 3G IMS dom=
ain.
 > > > > In particular in the IMS, I think that to provide multilmedia
 > > > > service (i.e video on demand), one can estabilish a=20
 > > > subscription to the service by SIP and then activate the=20
 > > > service with RTSP by DIGEST autentication with the=20
 > > > > parameters exchanged early during the subscription=20
 > > > (username and password)=20
 > > > >=20
 > > > > What do you think about that?
 > > > >=20
 > > > > Best Regards=20
 > > > >=20
 > > > > ----- Original Message -----=20
 > > > > From: "NAKAMURA, Hidefumi" <nakamura.hidefumi@lab.ntt.co.jp>
 > > > > To: <sip@ietf.org>; <mmusic@ietf.org>
 > > > > Cc: <hidefumi.nakamura@staff.east.ntt.co.jp>
 > > > > Sent: Monday, July 07, 2003 9:49 AM
 > > > > Subject: [Sip] Appropriate usage of SIP and RTSP
 > > > >=20
 > > > >=20
 > > > >=20
 > > > >>Hi, all.
 > > > >>
 > > > >>I have a question about the relationship between SIP and RTSP.=
=20
 > > > >>
 > > > >>In RFC3261, there is a description that says=20
 > > > >>   "SIP is not a vertically integrated communications=20
 > > > system.  SIP is
 > > > >>   rather a component that can be used with other IETF=20
 > > protocols to
 > > > >>   build a complete multimedia architecture.  Typically, these=

 > > > >>   architectures will include protocols such as the=20
 > > > Real-time Transport
 > > > >>   Protocol (RTP) (RFC 1889 [28]) for transporting=20
 > > > real-time data and
 > > > >>   providing QoS feedback, the Real-Time streaming protocol=20=

 > > > (RTSP) (RFC
 > > > >>   2326 [29]) for controlling delivery of streaming media, ...=
"
 > > > >>   (RFC3261 Section 2)
 > > > >>
 > > > >>However, my understanding is that both SIP and RTSP have=20
 > > > the capability
 > > > >>for "discovery of UAs or media streams."
 > > > >>If I use SIP to establish a stream session and use RTSP to con=
trol
 > > > >>the stream, there will be redundant sequences, i.e.=20
 > > > INVITE-OK-ACK and
 > > > >>SETUP-OK.
 > > > >>
 > > > >>Was there any discussion on how to use these two protocols=20
 > > > appropriately?
 > > > >>
 > > > >>With SIP application server, network providers can provide var=
ious
 > > > >>services on contents delivery while we will not be able to=20
 > > > do so only
 > > > >>with RTSP, I think.  On the other hand, RTSP has enough=20
 > > > capability for
 > > > >>discovery of media streams and for negotiation of SDPs.
 > > > >>So, I think, for the simple delivery of contents, RTSP is=20
 > > > appropriate to
 > > > >>use, and for various session control services regarding conten=
ts
 > > > >>delivery, SIP with RTSP is appropriate to use.
 > > > >>
 > > > >>Any opinions?
 > > > >>
 > > > >>I wonder why no contents delivery server seems to prepare=20
 > > > SIP stack now.=20
 > > > >>Is it because, up to now, there exists only a simple=20
 > > > contents delivery
 > > > >>service?
 > > > >>
 > > > >>best regards,
 > > > >>
 > > > >>--=20
 > > > >>NAKAMURA, Hidefumi=20
 > > > >>NTT ServiceIntegration /NetworkServiceSystems Labs.
 > > > >>tel: +81(422)59 3904; fax:+81(422)60 4012
 > > > >>---
 > > > >>
 > > > >>
 > > > >>
 > > > >>_______________________________________________
 > > > >>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=20
 > > application of sip
 > > > >>
 > > > >=20
 > > > >=20
 > > > >=20
 > > > > ---
 > > > > Outgoing mail is certified Virus Free.
 > > > > Checked by AVG anti-virus system (http://www.grisoft.com).
 > > > > Version: 6.0.495 / Virus Database: 294 - Release Date:=20
 > > > 30/06/2003J*fj)bz=09b=B2=D8m=B6>?=FF=0C> > 0=D6'=AD~S=E0=FEf=A2-=
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=CA'=B6=17s=A6(tm)bq=ABb=A2z=1F
 > > > >=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=20
 > > application of sip
 > > > >=20
 > > >=20
 > > >=20
 > > > --=20
 > > >=20
 > > > Magnus Westerlund
 > > >=20
 > > > Multimedia Technologies, Ericsson Research EAB/TVA/A
 > > >=20
 > > ------------------------------------------------------------------=
----
 > > > Ericsson AB                | Phone +46 8 4048287
 > > > Torshamsgatan 23           | Fax   +46 8 7575550
 > > > S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.=
com
 > > >=20
 > > >=20
 > > > _______________________________________________
 > > > mmusic mailing list
 > > > mmusic@ietf.org
 > > > https://www1.ietf.org/mailman/listinfo/mmusic
 > > >=20
 > >=20
 > > _______________________________________________
 > > mmusic mailing list
 > > mmusic@ietf.org
 > > https://www1.ietf.org/mailman/listinfo/mmusic
 > >=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

_______________________________________________
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: Robert Sparks <rsparks@dynamicsoft.com>
To: sip-implementors@cs.columbia.edu
Cc: sip@ietf.org, simple@ietf.org
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Subject: [Sip] Re: [Sip-implementors] SIPIT 13 Registration
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Registration for SIPIT 13 closes next week. If you
have not already registered, please do so now.

Thanks,
RjS

On Thu, 2003-06-26 at 14:53, Robert Sparks wrote:
> Registration for SIPIT 13 is open.
> 
> The event will be hosted by Mitel in the brand-new
> Brookstreet Hotel and Conference Center in Ottawa
> August 18-24th.
> 
> Information about the event and facilities as well
> as online registration can be found at:
> http://www.mitel.com/sipit/index.cfm
> 
> Registration through the website will close on July 18th.
> 
> See you there,
> RjS
> 
> 
> _______________________________________________
> Sip-implementors mailing list
> Sip-implementors@cs.columbia.edu
> http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors


_______________________________________________
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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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FYI: there is a sip2jabber service that connects SIP users to jabber chat rooms 
in the upcoming IETF meeting. More information can be found 
at http://www.iptel.org/ietf54/

-jiri

--
Jiri Kuthan            http://iptel.org/~jiri/ 


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At 10:38 PM 7/9/2003, Jiri Kuthan wrote:
>FYI: there is a sip2jabber service that connects SIP users to jabber chat rooms 
>in the upcoming IETF meeting. More information can be found 
>at http://www.iptel.org/ietf54/

typo: correct address is http://www.iptel.org/ietf57/

sorry,

-jiri 


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



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Reply-To: <vkenneally@cit.ie>
From: "Valerie Kenneally" <vkenneally@cit.ie>
To: <sip@ietf.org>
Date: Fri, 11 Jul 2003 12:01:35 +0100
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Subject: [Sip] SIP, Presence & Instant messaging
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Hi All,

	Does anyone know of any document(s) which describe exactly why SIP is
suitable for IM & P?
Cheers,
Valerie Kenneally


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Subject: Re: [MMUSIC] Re: [Sip] Appropriate usage of SIP and RTSP
From: Cullen Jennings <fluffy@cisco.com>
To: Brian Rosen <Brian.Rosen@marconi.com>,
        "'Yutaka Takeda'" <takeday@kmerl.com>, <sip@ietf.org>,
        <mmusic@ietf.org>
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In many ways, this problem is similar to the far end camera control issues.
Real time responses is needed, It is a 1:1 relationship between the
controller and controlee, the control protocol needs to be extensible.

On 7/9/03 11:40, "Rosen, Brian" <Brian.Rosen@marconi.com> wrote:

> I've been thinking for a long time about what an appropriate
> mechanism to provide a voice/videomail client on a sipphone
> would be.  It's really clear that you record and play messages
> with VCR like functionality.  For that, you need start/stop/pause/
> rewind/fast forward, and so I was thinking RTSP.  On the other
> hand, you surely establish the session to the voicemail server
> with a sip call.
>=20
> Thus, I like the way Jonathan is thinking, but you also need
> message selection/creation/deletion functions, and for those
> I was thinking KPML, and then you wonder if you just use that
> as the signalling mechanism for the record/play, and I get
> confused.
>=20
> Brian
>=20
>> -----Original Message-----
>> From: Yutaka Takeda [mailto:takeday@kmerl.com]
>> Sent: Wednesday, July 09, 2003 2:11 PM
>> To: sip@ietf.org; mmusic@ietf.org
>> Subject: RE: [MMUSIC] Re: [Sip] Appropriate usage of SIP and RTSP
>>=20
>>=20
>> The applications that have need for SIP and RTSP features I
>> can imagine are:
>>=20
>> - Voice&video mail (similar to 'sipum' but VM server is in
>> the SIP phone)
>> - Personal broadcast station from my PC at home.
>> - Watching video recorded with VCR at home or office from remote...
>>=20
>> What I am not sure is if RTSP has any solution or a framework for a
>> RTSP server behind a NAT. If not, the solution is likely to
>> be a repeat
>> of what SIP provides. I thought TURN could be used in order
>> for signaling
>> message to reach RTSP server behind a NAT, but I noticed that
>> it is not=20
>> applicable (purposefully) for a server running behind a NAT because of
>> a security consideration. Lots of issues for RTSP in this area alone?
>>=20
>> I believe a lot of people are interested in this discussion.
>>=20
>> Best regards,
>> Yutaka
>>=20
>>=20
>>> -----Original Message-----
>>> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
>>> Sent: Wednesday, July 09, 2003 5:53 AM
>>> To: Pianigiani, Jacopo (Jacopo)
>>> Cc: sip@ietf.org; mmusic@ietf.org
>>> Subject: [MMUSIC] Re: [Sip] Appropriate usage of SIP and RTSP
>>>=20
>>>=20
>>> Hi,
>>>=20
>>> When it comes to IMS and 3G terminals, most will already
>>> before having=20
>>> an IMS functionality have support for PSS. PSS is
>>> Packet-based Streaming
>>> Service which is an umbrella kind of standard for streaming service
>>> defined in 3GPP targeted at low bit-rates and mobile networks.
>>> Specification available from Rel 4 in TS 26.233 and TS
>>> 26.234. Already
>>> today there exist GSM/GPRS phones that has a PSS client.
>>>=20
>>> So I think that dual stack implementations will not be that
>>> impossible=20
>>> to foresee.
>>>=20
>>> However I think the main issue in this discussion is to
>> look at what=20
>>> applications that has need for SIP-RTSP interactions. Based
>>> on this we=20
>>> can see if there is solutions easily available or not.
>>>=20
>>> I can agree with Rosenberg that if we where going to design a RTSP
>>> functionality protocol today, looking at SIP for session
>>> establishment=20
>>> would be interesting. However that opportunity has passed
>> as RTSP has=20
>>> started to be deployed in increasing numbers. I think even
>> the latest=20
>>> Windows media player does support RTSP.
>>>=20
>>>=20
>>> Best Regards
>>>=20
>>> Magnus Westerlund
>>>=20
>>>=20
>>> Pianigiani, Jacopo (Jacopo) wrote:
>>>> Hello all,
>>>> In my undertanding RTSP is suitable for broadcast, unicast
>>> or multicast streams distribution. The RTSP protocol was
>>> ideated for typical content delivery, and in effect for this
>>> application could be more than suitable. But referring to an
>>> IMS domain, in my point of view adopting RSTP as media
>>> streams control, would imply adopting at least a double
>>> protocol stack in the terminal equipment. The IMS concept was
>>> mainly driven by the concept - in my understanding - of
>>> simplifying the terminal (making it simpler from a protocol
>>> stack implementation means lowering costs and increasing
>>> penetration on the market) by adopting a single protocol
>>> stack in the terminal, and having the RAN and the core
>>> evolving toward a unified transport mechanism for which an
>>> almost unified level of control on the network resources
>>> allocated to a session (whatever session we refer to, voice ,
>>> multimedia p-t-t call, web browsing, multimedia session with
>>> voice call and dynamic web pages).
>>>> So in this sense, my understanding is that the RTSP could
>>> be suitable for media content, but it would imply again new
>>> complexity on the IMS UMTS/CDMA terminal itself, which is
>>> against the idea of IMS of enhancing the services the user
>>> can access by having an homogeneous protocol stack on the
>>> terminal itself for all types of services.
>>>>=20
>>>> After all, the RTSP , apart from content delivery (of
>>> streams), could not definitely handle appropriately sessions
>>> like Voice calls + Interactive Web Pages.
>>>> Regards
>>>>=20
>>>> Jacopo Pianigiani
>>>> Lucent Technologies - Italy
>>>> =E839 335 7350536
>>>>=20
>>>> -----Original Message-----
>>>> From: Paolo Asprino [mailto:asprino@coritel.it]
>>>> Sent: Wednesday, July 09, 2003 9:40 AM
>>>> To: NAKAMURA, Hidefumi; sip@ietf.org; mmusic@ietf.org
>>>> Subject: Re: [Sip] Appropriate usage of SIP and RTSP
>>>>=20
>>>>=20
>>>> Hi all
>>>>=20
>>>> I'm thinking a solution to join SIP and RTSP in the 3G IMS domain.
>>>> In particular in the IMS, I think that to provide multilmedia
>>>> service (i.e video on demand), one can estabilish a
>>> subscription to the service by SIP and then activate the
>>> service with RTSP by DIGEST autentication with the
>>>> parameters exchanged early during the subscription
>>> (username and password)
>>>>=20
>>>> What do you think about that?
>>>>=20
>>>> Best Regards=20
>>>>=20
>>>> ----- Original Message -----
>>>> From: "NAKAMURA, Hidefumi" <nakamura.hidefumi@lab.ntt.co.jp>
>>>> To: <sip@ietf.org>; <mmusic@ietf.org>
>>>> Cc: <hidefumi.nakamura@staff.east.ntt.co.jp>
>>>> Sent: Monday, July 07, 2003 9:49 AM
>>>> Subject: [Sip] Appropriate usage of SIP and RTSP
>>>>=20
>>>>=20
>>>>=20
>>>>> Hi, all.
>>>>>=20
>>>>> I have a question about the relationship between SIP and RTSP.
>>>>>=20
>>>>> In RFC3261, there is a description that says
>>>>>   "SIP is not a vertically integrated communications
>>> system.  SIP is
>>>>>   rather a component that can be used with other IETF
>> protocols to
>>>>>   build a complete multimedia architecture.  Typically, these
>>>>>   architectures will include protocols such as the
>>> Real-time Transport
>>>>>   Protocol (RTP) (RFC 1889 [28]) for transporting
>>> real-time data and
>>>>>   providing QoS feedback, the Real-Time streaming protocol
>>> (RTSP) (RFC
>>>>>   2326 [29]) for controlling delivery of streaming media, ..."
>>>>>   (RFC3261 Section 2)
>>>>>=20
>>>>> However, my understanding is that both SIP and RTSP have
>>> the capability
>>>>> for "discovery of UAs or media streams."
>>>>> If I use SIP to establish a stream session and use RTSP to control
>>>>> the stream, there will be redundant sequences, i.e.
>>> INVITE-OK-ACK and
>>>>> SETUP-OK.
>>>>>=20
>>>>> Was there any discussion on how to use these two protocols
>>> appropriately?
>>>>>=20
>>>>> With SIP application server, network providers can provide various
>>>>> services on contents delivery while we will not be able to
>>> do so only
>>>>> with RTSP, I think.  On the other hand, RTSP has enough
>>> capability for
>>>>> discovery of media streams and for negotiation of SDPs.
>>>>> So, I think, for the simple delivery of contents, RTSP is
>>> appropriate to
>>>>> use, and for various session control services regarding contents
>>>>> delivery, SIP with RTSP is appropriate to use.
>>>>>=20
>>>>> Any opinions?
>>>>>=20
>>>>> I wonder why no contents delivery server seems to prepare
>>> SIP stack now.=20
>>>>> Is it because, up to now, there exists only a simple
>>> contents delivery
>>>>> service?
>>>>>=20
>>>>> best regards,
>>>>>=20
>>>>> --=20
>>>>> NAKAMURA, Hidefumi
>>>>> NTT ServiceIntegration /NetworkServiceSystems Labs.
>>>>> tel: +81(422)59 3904; fax:+81(422)60 4012
>>>>> ---
>>>>>=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
>>>>=20
>>>>=20
>>>>=20
>>>> ---
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>>>> Checked by AVG anti-virus system (http://www.grisoft.com).
>>>> Version: 6.0.495 / Virus Database: 294 - Release Date:
>>> 30/06/2003J*fj)bz    b=B2=D8m=B6>?=FF=0C> > 0=D6'=AD~S=E0=FEf=A2-f=A7=FEX=AC=B6)=DF=A3=FB"=A58b=B2X=AC=B6+=1F=A2=B3DY=D7=AF
>>> zZ)(tm)=E9=ED=A1=FBay=CA+y"=0F>=BA-=A1=CA%R=C7=ACS~=A6=A6W=A6z{h=AE=C7,r?n(tm)=B8sy=DBY=A2=BA=AEz=CBb=A2{(=9D=CB=AB
>>> =AD=E9=ED=B2*T=B1=EB"=A6~=A7''=AD~S=E0~S=E7{=07^=BD=E9h=A6g=A7=B6=CA'=B6=17s=A6(tm)bq=ABb=A2z=1F
>>>>=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
>>>=20
>>>=20
>>> --=20
>>>=20
>>> Magnus Westerlund
>>>=20
>>> Multimedia Technologies, Ericsson Research EAB/TVA/A
>>>=20
>> ----------------------------------------------------------------------
>>> Ericsson AB                | Phone +46 8 4048287
>>> Torshamsgatan 23           | Fax   +46 8 7575550
>>> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>>>=20
>>>=20
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www1.ietf.org/mailman/listinfo/mmusic
>>>=20
>>=20
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www1.ietf.org/mailman/listinfo/mmusic
>>=20
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
>=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



From exim@www1.ietf.org  Mon Jul 14 05:20:37 2003
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From: "Jesske, R" <R.Jesske@telekom.de>
To: gonzalo.camarillo@ericsson.com
Cc: sip@ietf.org
Date: Mon, 14 Jul 2003 09:53:11 +0200
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Subject: [Sip] Reason Header RFC 3326
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Hi Gonzalo et all,
I have a question with regard to the Reason Header RFC3326.
In RFC 3326 the Reason header includes a reason-text parameter element. In combination with the Q.850 Cause Value which text should be in the "reason-text" parameter?
Should it be the Q.850 definition text or could it be something else?

Best Regards

Roland


Deutsche Telekom AG
T Com Zentrale
Roland Jesske, T38-12
Section T38; Signalling, Gateways and Switching Systems 
Am Kavalleriesand 3, 64295 Darmstadt, Germany
Phone:  +49 6151 83-5940 
Fax:      +49 6151 83-4577 
email:   r.jesske@telekom.de




_______________________________________________
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 exim@www1.ietf.org  Mon Jul 14 06:00:46 2003
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Date: Wed, 09 Jul 2003 17:11:24 -0400
From: "C. Wegrzyn" <wegrzyn@garbagedump.com>
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To: Jiri Kuthan <jiri@iptel.org>
CC: sip@ietf.org, sipping@ietf.org
References: <5.2.0.9.0.20030709222843.0307c5f8@localhost>
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Subject: [Sip] Re: [Sipping] text conferencing at ietf57
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I don't see anything at the provided URL.

Chuck Wegrzyn


Jiri Kuthan wrote:

>FYI: there is a sip2jabber service that connects SIP users to jabber chat rooms 
>in the upcoming IETF meeting. More information can be found 
>at http://www.iptel.org/ietf54/
>
>-jiri
>
>--
>Jiri Kuthan            http://iptel.org/~jiri/ 
>
>
>_______________________________________________
>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



From exim@www1.ietf.org  Mon Jul 14 06:00:50 2003
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content-class: urn:content-classes:message
Subject: RE: [MMUSIC] Re: [Sip] Appropriate usage of SIP and RTSP
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Thread-Topic: [MMUSIC] Re: [Sip] Appropriate usage of SIP and RTSP
Thread-Index: AcNGLV4de+FJnurnRWaKHIBOOZMeaQAEiCBA
From: "Yutaka Takeda" <takeday@kmerl.com>
To: <sip@ietf.org>, <mmusic@ietf.org>
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The applications that have need for SIP and RTSP features I
can imagine are:

- Voice&video mail (similar to 'sipum' but VM server is in the SIP =
phone)
- Personal broadcast station from my PC at home.
- Watching video recorded with VCR at home or office from remote...

What I am not sure is if RTSP has any solution or a framework for a
RTSP server behind a NAT. If not, the solution is likely to be a repeat
of what SIP provides. I thought TURN could be used in order for =
signaling
message to reach RTSP server behind a NAT, but I noticed that it is not=20
applicable (purposefully) for a server running behind a NAT because of
a security consideration. Lots of issues for RTSP in this area alone?

I believe a lot of people are interested in this discussion.

Best regards,
Yutaka


> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Wednesday, July 09, 2003 5:53 AM
> To: Pianigiani, Jacopo (Jacopo)
> Cc: sip@ietf.org; mmusic@ietf.org
> Subject: [MMUSIC] Re: [Sip] Appropriate usage of SIP and RTSP
>=20
>=20
> Hi,
>=20
> When it comes to IMS and 3G terminals, most will already=20
> before having=20
> an IMS functionality have support for PSS. PSS is=20
> Packet-based Streaming=20
> Service which is an umbrella kind of standard for streaming service=20
> defined in 3GPP targeted at low bit-rates and mobile networks.=20
> Specification available from Rel 4 in TS 26.233 and TS=20
> 26.234. Already=20
> today there exist GSM/GPRS phones that has a PSS client.
>=20
> So I think that dual stack implementations will not be that=20
> impossible=20
> to foresee.
>=20
> However I think the main issue in this discussion is to look at what=20
> applications that has need for SIP-RTSP interactions. Based=20
> on this we=20
> can see if there is solutions easily available or not.
>=20
> I can agree with Rosenberg that if we where going to design a RTSP=20
> functionality protocol today, looking at SIP for session=20
> establishment=20
> would be interesting. However that opportunity has passed as RTSP has=20
> started to be deployed in increasing numbers. I think even the latest=20
> Windows media player does support RTSP.
>=20
>=20
> Best Regards
>=20
> Magnus Westerlund
>=20
>=20
> Pianigiani, Jacopo (Jacopo) wrote:
> > Hello all,
> > In my undertanding RTSP is suitable for broadcast, unicast=20
> or multicast streams distribution. The RTSP protocol was=20
> ideated for typical content delivery, and in effect for this=20
> application could be more than suitable. But referring to an=20
> IMS domain, in my point of view adopting RSTP as media=20
> streams control, would imply adopting at least a double=20
> protocol stack in the terminal equipment. The IMS concept was=20
> mainly driven by the concept - in my understanding - of=20
> simplifying the terminal (making it simpler from a protocol=20
> stack implementation means lowering costs and increasing=20
> penetration on the market) by adopting a single protocol=20
> stack in the terminal, and having the RAN and the core=20
> evolving toward a unified transport mechanism for which an=20
> almost unified level of control on the network resources=20
> allocated to a session (whatever session we refer to, voice ,=20
> multimedia p-t-t call, web browsing, multimedia session with=20
> voice call and dynamic web pages).
> > So in this sense, my understanding is that the RTSP could=20
> be suitable for media content, but it would imply again new=20
> complexity on the IMS UMTS/CDMA terminal itself, which is=20
> against the idea of IMS of enhancing the services the user=20
> can access by having an homogeneous protocol stack on the=20
> terminal itself for all types of services.
> >=20
> > After all, the RTSP , apart from content delivery (of=20
> streams), could not definitely handle appropriately sessions=20
> like Voice calls + Interactive Web Pages.=20
> > Regards
> >=20
> > Jacopo Pianigiani
> > Lucent Technologies - Italy
> > =E839 335 7350536
> >=20
> > -----Original Message-----
> > From: Paolo Asprino [mailto:asprino@coritel.it]
> > Sent: Wednesday, July 09, 2003 9:40 AM
> > To: NAKAMURA, Hidefumi; sip@ietf.org; mmusic@ietf.org
> > Subject: Re: [Sip] Appropriate usage of SIP and RTSP
> >=20
> >=20
> > Hi all
> >=20
> > I'm thinking a solution to join SIP and RTSP in the 3G IMS domain.
> > In particular in the IMS, I think that to provide multilmedia
> > service (i.e video on demand), one can estabilish a=20
> subscription to the service by SIP and then activate the=20
> service with RTSP by DIGEST autentication with the=20
> > parameters exchanged early during the subscription=20
> (username and password)=20
> >=20
> > What do you think about that?
> >=20
> > Best Regards=20
> >=20
> > ----- Original Message -----=20
> > From: "NAKAMURA, Hidefumi" <nakamura.hidefumi@lab.ntt.co.jp>
> > To: <sip@ietf.org>; <mmusic@ietf.org>
> > Cc: <hidefumi.nakamura@staff.east.ntt.co.jp>
> > Sent: Monday, July 07, 2003 9:49 AM
> > Subject: [Sip] Appropriate usage of SIP and RTSP
> >=20
> >=20
> >=20
> >>Hi, all.
> >>
> >>I have a question about the relationship between SIP and RTSP.=20
> >>
> >>In RFC3261, there is a description that says=20
> >>   "SIP is not a vertically integrated communications=20
> system.  SIP is
> >>   rather a component that can be used with other IETF protocols to
> >>   build a complete multimedia architecture.  Typically, these
> >>   architectures will include protocols such as the=20
> Real-time Transport
> >>   Protocol (RTP) (RFC 1889 [28]) for transporting=20
> real-time data and
> >>   providing QoS feedback, the Real-Time streaming protocol=20
> (RTSP) (RFC
> >>   2326 [29]) for controlling delivery of streaming media, ..."
> >>   (RFC3261 Section 2)
> >>
> >>However, my understanding is that both SIP and RTSP have=20
> the capability
> >>for "discovery of UAs or media streams."
> >>If I use SIP to establish a stream session and use RTSP to control
> >>the stream, there will be redundant sequences, i.e.=20
> INVITE-OK-ACK and
> >>SETUP-OK.
> >>
> >>Was there any discussion on how to use these two protocols=20
> appropriately?
> >>
> >>With SIP application server, network providers can provide various
> >>services on contents delivery while we will not be able to=20
> do so only
> >>with RTSP, I think.  On the other hand, RTSP has enough=20
> capability for
> >>discovery of media streams and for negotiation of SDPs.
> >>So, I think, for the simple delivery of contents, RTSP is=20
> appropriate to
> >>use, and for various session control services regarding contents
> >>delivery, SIP with RTSP is appropriate to use.
> >>
> >>Any opinions?
> >>
> >>I wonder why no contents delivery server seems to prepare=20
> SIP stack now.=20
> >>Is it because, up to now, there exists only a simple=20
> contents delivery
> >>service?
> >>
> >>best regards,
> >>
> >>--=20
> >>NAKAMURA, Hidefumi=20
> >>NTT ServiceIntegration /NetworkServiceSystems Labs.
> >>tel: +81(422)59 3904; fax:+81(422)60 4012
> >>---
> >>
> >>
> >>
> >>_______________________________________________
> >>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
> >=20
> >=20
> > ---
> > Outgoing mail is certified Virus Free.
> > Checked by AVG anti-virus system (http://www.grisoft.com).
> > Version: 6.0.495 / Virus Database: 294 - Release Date:=20
> 30/06/2003J*fj)bz	b=B2=D8m=B6>?=FF=0C> =
0=D6'=AD~S=E0=FEf=A2-f=A7=FEX=AC=B6)=DF=A3=FB"=A58b=B2X=AC=B6+=1F=A2=B3DY=
=D7=AF
> =
zZ)(tm)=E9=ED=A1=FBay=CA+y"=0F>=BA-=A1=CA%R=C7=ACS~=A6=A6W=A6z{h=AE=C7,r?=
n(tm)=B8sy=DBY=A2=BA=AEz=CBb=A2{(=9D=CB=AB
> =
=AD=E9=ED=B2*T=B1=EB"=A6~=A7''=AD~S=E0~S=E7{=07^=BD=E9h=A6g=A7=B6=CA'=B6=17=
s=A6(tm)bq=ABb=A2z=1F
> >=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
>=20
>=20
> --=20
>=20
> Magnus Westerlund
>=20
> Multimedia Technologies, Ericsson Research EAB/TVA/A
> ----------------------------------------------------------------------
> Ericsson AB                | Phone +46 8 4048287
> Torshamsgatan 23           | Fax   +46 8 7575550
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>=20
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
>=20

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Date: Tue, 15 Jul 2003 20:58:30 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Subject: [Sip] comments on history-info-00
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Mary,

Here are some comments on history-info. I suspect many apply to prior 
versions as well. My apologies if they have been discussed, I have not 
really followed the work closely. Many of my comments are asking for 
much more precision in the document. Generally, I found it to be very 
imprecise, with lots of undefined terminology and vague procedures.

That aside, I have some major issues with the privacy mechanisms, with 
the usage of history-info by redirect servers, with sending a 183 with 
history information, and with the backwards compatibility 
implications. So, here we go:



Firstly, the draft says in many places that applications need to 
document the impact of things. For example, Section 1.1 says:

    Thus, it is highly
    recommended that all applications making use of the request history
    information clearly define the impact of the information not being
    available and specify the processing of such a request.

Does this mean there is some kind of application registry where such 
documentation will exist? Does it mean that applications won't be 
interoperable because they will depend on the presence of these 
headers? The text in 1.1 makes it sound like the information would 
only be present if some local policy prevented its insertion. What 
about cases where the proxy doesn't support it? Even if it does, your 
draft reads very much like it is now going to require every proxy to 
insert this header. Is that so?


Second, section 1.3 talks about the use of the Privacy header. There 
are a few problems here. First, the privacy header gets stripped once 
the privacy service finishes execution. Thus, a downstream proxy would 
not be able to know privacy was requested. To a large degree, this is 
because the privacy header in a request is about privacy of the 
caller. Request history is about the current target (i.e., called 
party). So, in fact, the issue is that if the *called* party requests 
privacy (using a privacy header in a response), the history info 
headers should not be propagated backwards in a response. I dont think 
your draft does that.

Local policy might also play a role, when it is representing the 
privacy interests of a user that is retargeting. In that case, it was 
unclear to me how the proxy prevents downstream elements from 
inserting history-info? Your draft makes it sound like the privacy 
requirements are met if just this one hop avoids insertion. But seeing 
the contact from downstream retargets will often reveal sensitive 
information about a target further upstream.


Section 2.1:

> Targeted-to-URI: the Request URI captured as the Request is 
>         targeted. By capturing a copy of the Request URI in the initial 
>         request, the Retargeted-from-URI is already captured when a 
>         request is retargeted and the Retargeted-to-URI is being 
>         captured.   
>  

I read this many times and could not make sense of it. What does it mean?

Section 2.3.1:

> The UAC SHOULD include the HistInfo option tag in the Supported 
>    header in any request not associated with an established dialog for 
>    which the UAC would like the History-Info in the Response.  In 
>    addition, the UAC should initiate the capturing of the History 
>    Information by capturing the Request-URI as the hi-targeted-to-uri 
>    and initializing the index to 1.  

What if the UAC doesn't support this extension? Can the UAS get 
history info? My understanding is that most, if not all, evnisioned 
applications were for the UAS. This means the UAS can't get 
information unless the caller and many, if not all, proxies along the 
path support it.

Section 2.3.2:

> The processing of History-Info by a UAS in a Request depends upon 
>    local policy and specific applications at the UAS which might make 
>    use of the information.  If the HistInfo option tag is received in a 
>    request, the UAS should include any History-Info received in the 
>    request in the subsequent response.     

I think you mean SHOULD in the last sentence. Also, I think you mean 
that "If the request contained the histinfo option tag in the 
Supported header field of the request, the UAS....".

Also, you've capitalized the H and I in HistInfo. Please don't. Option 
tags are not case sensitive.

Section 2.3.3:

The wording about when to capture is really vague. You need to be much 
more precise in your wording of when the proxy is and isnt going to 
capture history information.

Section 2.3.3.1:

>  If the proxy supports History-Info, the proxy SHOULD add any History-
>    Info collected as it retargets a Request. For retargets that are the 
>    result of an explicit SIP response, the SIP Response Code that 
>    triggered the retargeting MUST be included in the Reason header of 
>    the Targeted-to-URI.

This needs to be more precise. What does it mean "..collected as it 
retargets a request"? How is this information collected? Also, when 
you say "explicit SIP response" - all responses are explicit as far as 
I know. Perhaps you mean when a 3xx response is received? Also, you 
say "Reason header of the Targeted-to-URI". I believe reason is a 
parameter, not a header, in this instance.

and then:
> For retargets as a result of timeouts or 
>    internal events, a Reason header MAY be included in the Reason header 
>    of the Targeted-to-URI. 

what are internal events?

What value does the reason parameter have in the case of timeout? In 
the case of internal events?

and then:
>  Additionally, if a request is received that doesn't include a 
>    captured Request URI from the previous entity, the proxy MAY add an 
>    additional entry, effectively capturing the retargeted-from-URI in 
>    the Request.   


What does it mean for a request to "not include a captured 
Request-URI"? You need to define that. What is contained in the 
additional entry?

I may sound like I am being nit-picky, but in my experience 
specifications need to be very precise in what you want people to do. 
I am confused about what is implied here, and I would imagine many 
others would be confused too.

and then:

> In order to maintain ordering and accurately reflect the nesting and 
>    retargeting of the request, an index MUST be included along with the 
>    Targeted-to-URI being captured. The basic rule for adding the index 
>    are to read the value from the previous History-Info, if available, 
>    and capture the index.n as the index for the History-Info being 
>    captured, where n would typically be 1 for a forwarded request. Thus, 
>    the level of nesting of the index reflects the number of hops.

define previous.

what does it mean to "capture the index.n"? what is index? what is .n? 
  You say its "typically 1". When is it 1, and when is it not one?

and then:
> For 
>    retargets within a proxy, the proxy MUST maintain the current level 
>    of nesting by incrementing the lowest/last digit of the index for 
>    each instance of retargeting, thus reflecting the number of retargets 
>    within the proxy.

I cannot understand this at all.

and then:
> n index MUST NOT be added 
>    in the scenario whereby the received request had no History-Info 
>    header and the retargeted-from-URI is being captured for 
>    completeness.

what does "for completeness" mean? You need to be precise.

> The lack of Reason headers in the captured Request-URIs should be 
>    indicative of the parallel nature of forking (i.e the Request-URIs 
>    are not the result of retargets, but are rather all simultaneous 
>    Targeted-To URIs.)  

should be indicative? Is it, or isnt it?

> A proxy that receives a Request with the HistInfo option tag in the 
>    Supported header, and depending upon a local policy supporting the 
>    capture of History-Info, SHOULD return captured History-Info in 
>    subsequent, provisional and final responses to the Request.  A 183 
>    response MAY be sent explicitly for the purposes of conveying 
>    History-Info prior to the final response. 

I think the first sentence means that the proxy does nothing to 
responses - any history-info in those responses is just passed on. But 
I'm not sure. The sentence says "should return", and return seems like 
an active operation to me. The next sentence actually implies that the 
proxy generate a 183 provisional responses. This means that a UA 
making a call with N proxies will get N extra provisional responses, 
where the first has one history-info value, the next 2, the next 3, 
and so on. The first history info is transmitted to the UAC N times. 
Is that really what we want? It seems very expensive for the UAC.


> It MAY be advantageous for redirect servers to support the receipt of 
>    History-Info in requests.

You can you assign a protocol strength to being advantageous?

I think you mean "A redirect server MAY use information in the 
History-Info header field in a request to determine the URIs that it 
will retarget to.".

> By receiving it in the request, the 
>    Redirect Server MAY be able to optimize the information it sends in 
>    responses by looking at the already targeted-to-URIs.

and what optimization is that? I think you mean that the redirect 
server shouldnt redirect to somewhere that the request has already 
visited. If so, please say that. However, are we sure about that? 
Redirection to a URI already visited can be appropriate in cases where 
there is a spiral, and not a loop. Unless a redirect server is 
maintaining state, how can it make this determiantionabout whether 
this is a spiral or a loop?



> HistInfo      When used with the Supported header, [RFCXXXX] 
>                  this option tag indicates support 
>                  for the History Information to be  
>                  captured for requests and returned in 
>                  subsequent responses. This tag is not 
>                  used in a Proxy-Require or Requires  
>                  header field since support of  
>                  History-Info is optional.       


Require header field, not Requires.


Thanks,
Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com




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From exim@www1.ietf.org  Wed Jul 16 04:18:51 2003
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Date: Wed, 16 Jul 2003 04:15:54 -0400
From: "David R. Oran" <oran@cisco.com>
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Subject: [Sip] Treatment of q values on contacts returned in redirects
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The proposal Jonathan made on this is to effectively replace the contact 
that returned the redirect with the contacts in the redirect, and then try 
them in the order of their q values - essentially a depth-first search.

In the SIP meeting, I questioned that this was too restrictive and that 
there was no need to actually mandate that behavior. That is one 
possibility, but not the only one.

As an alternative, what one could do is to go back the the original set of 
registered contacts, eliminate the ones already tried and the one that 
returned the redirect, add the ones in the redirect response and treat them 
as if they were directly registered contacts for the AoR. You throw away 
the results of the current caller-prefs calculation, and rerun it from 
scratch on this set of contacts.

Why is this alternative needed? Imagine a contact which does "Call forward 
no answer" by timing out an alert state and sends back a redirect to some 
random place before the upstream proxy gives up.

What if the callee's policy is to try other of his registered contacts 
first before inviting to a CFNA number returned by a contact?

Sorry for the terseness of this - wanted to get this out right away before 
I forget what the issue is.

Dave.

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Cc: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
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Subject: [Sip] Comment Period on Callee Capabilities
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Hi All,

After the split between caller preferences and callee capabilities, we 
need to have a short review period on the callee capabilities draft 
(draft-ietf-sip-callee-caps-00.txt).  This draft has already been 
through detailed review and 2 WGLCs as part of its original parent 
document, but we want to make sure that the draft holds together in its 
present form. There will be a separate thread (Jonathan will start this 
thread) about the possibility of adding a device-id to the draft. 
Please send your comments to the sip list or to authors. This comment 
period will close on Friday August 1st.

Barring catastrophe, the chairs would like to send this document to the 
IESG shortly after the review period expires, as many other documents 
(including several documents in other working groups  (IPTEL, SIMPLE, 
SIPPING)) depend on this document.

many thanks,
-rohan
co-chair SIP and SIPPING


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From exim@www1.ietf.org  Wed Jul 16 09:21:35 2003
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Date: Wed, 16 Jul 2003 16:19:10 +0300
From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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Subject: [Sip] Re: Reason Header RFC 3326
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Like the reason text in SIP responses, the reason text is intened for
human consumption. Therefore, you can put whatever you want there.

Gonzalo

"Jesske, R" wrote:
> 
> Hi Gonzalo et all,
> I have a question with regard to the Reason Header RFC3326.
> In RFC 3326 the Reason header includes a reason-text parameter element. In combination with the Q.850 Cause Value which text should be in the "reason-text" parameter?
> Should it be the Q.850 definition text or could it be something else?
> 
> Best Regards
> 
> Roland
> 
> Deutsche Telekom AG
> T Com Zentrale
> Roland Jesske, T38-12
> Section T38; Signalling, Gateways and Switching Systems
> Am Kavalleriesand 3, 64295 Darmstadt, Germany
> Phone:  +49 6151 83-5940
> Fax:      +49 6151 83-4577
> email:   r.jesske@telekom.de

-- 
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: Werkstudent8 SAL <werkstudent8.sal@mch.siemens.de>
To: "'sip@ietf.org'" <sip@ietf.org>
Subject: [SIP] Replaces header and REFER
Date: Wed, 16 Jul 2003 15:38:49 +0200
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From: "Stefan Mecke" <werkstudent8.sal@mch.siemens.de>

Hi all, 

I have a question about draft-ietf-sip-replaces-03.txt:
Section 2 says:
   ... the Replaces header is
   frequently used in combination with the REFER [4] method ...

but section 3 contains:
   ... if
   a Replaces header field is present in a request other than INVITE,
   the UAS MUST reject the request ..

I think it makes a lot of sense to allow replaces with REFER; e.g. in the
call park scenario you could have a web page with all call and make a "click
to retrieve from park" to have it on your desk phone.
And that is similar to what's described in
draft-ietf-sipping-cc-transfer-01.txt, although I have a notion that there
is some difference but I don't manage to get it out of the draft. 

Thanks

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From exim@www1.ietf.org  Wed Jul 16 11:19:35 2003
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Subject: [Sip] Device ID callee capabilitiy
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Folks,

One of the issues I raised today during the meeting was on the utility 
of the uri-user and uri-domain parameters in 
draft-ietf-sip-callee-caps. The primary purpose that these serve is a 
means of identifying a UA instance. In order to accomplish 
applicaitons like assisted call transfer, a UA would send INVITE to 
the AOR of the transfer target, and include an Accept-Contact with the 
uri-user and uri-domain equal to those of the UA instance.

Now, it turns out that uri-user and uri-domain are not very helpful 
when used this way. GRUU is a better approach. However, thats almost a 
secondary issue.

The primary issue is that uri-user and uri-domain are ugly, and they 
repeat information already in the registration. The suggestion, made 
by Paul, was to instead define a device-id attribute. This ID would be 
long lived - burned into firmware of a device, perhaps. It would be 
included in a registration as a unique ID for the device or software 
instance, as the case may be.

You could use this callee capability attribute for the assisted 
transfer just as you could use uri-user and uri-domain. It has the 
same problems that still make GRUU a better solution. However, during 
the meeting, folks like the idea of a device-id attribute as something 
useful for lots of other applications.

So, I would like to invite comment and discussion on (1) whether you 
like the idea of replacing the uri-user and uri-domain attributes with 
device-id (a better name may be needed, like instance-id), (2) what 
kind of usages you had in mind for it, for good or for evil.

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com



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From: "Dean Willis" <dean.willis@softarmor.com>
To: <sip@ietf.org>
Cc: <rohan@cisco.com>, <mankin@psg.com>
Date: Thu, 17 Jul 2003 07:42:35 -0500
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Subject: [Sip] Draft SIP Minutes, IETF 57
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Please review and send any needed corrections.

This text (or revisions thereof) will be maintained at:
http://www.softarmor.com/sipwg/meets/ietf57/notes/minutes.html


Draft Minutes,  SIPWG IETF57
Recorded by  Alan Johnston (alan.johnston@mci.com)
Edited by Dean Willis (dean.willis@softarmor.com)
Revised 17Jul2003
Meeting July 16, 2003,  0900

Start
-------
Meeting called to order by chairs.
Proposed agenda reviewed and accepted.
Announcement: Jon Peterson has retired as SIP co-chair.
Brian Rosen volunteered as chat room moderator.
Alan Johnston volunteered as scribe.

Status of Drafts - Chairs
---------------------------------
New Pubs: RFC 3515 REFER published

MIB: Volunteers to review a section of MIB: Orit Levin, Mary Barnes, =
Cullen
Jennings, and Kent (Rohan has his info). Rohan will assign some more
volunteers.

congest-safe:  Proposal for author to remove UDP kludge.  No objections =
or
discussion.



PUBLISH - Aki Niemi
------------------------------
Open Issue: collision recovery.  Proposed solution: query principle, may
subscribe to package. No objections or discussion.

Open Issue: PUBLISH and dialogs.  Proposed solution: add text about =
dialogs,
discourage reuse. No objections or discussion.

Open Issue: atomicity. Proposed solution: relax restriction about
overlapping requests.

Comment: Go with a simpler model - all tuples are independent.  Solve =
using
composition and authorization instead of publication.  Or, most recent
publisher overrides any others.

Comment: Agree with comment. Don't reinvent WebDAV.  Don't make endpoint
behavior too complex.=20

Comment: Agree with comment that this is an authorization problem.

Comment: Not clear we can relax overlap with congestion issues.

Conclusion: Author will take to the list for more discussion.


Resource Priority - Henning Schulzrinne
-------------------------------------------------------
Issue: error handling.  Proposed solution: 503 or 403 or 417 (Unknown
Resource Priority - only if Require is used. No objections or =
discussion.

Believed to be ready for WGLC

Comment: Pointers to name spaces are in draft.

Comment: Role based authorization is still moving forward.

Volunteers to review the next draft: Paul Kyzivat, Ben Campbell

Caller Prefs - Jonathan Rosenberg
-----------------------------------------------
Issue in Callee Caps - URI-user and URI-domain - duplication: Suggested  =
Use
a Device ID (Contact URI attribute) instead? No conclusion in this
discussion.

Comment: Device ID is interesting, but could be overloaded. Should =
recommend
GRUU for attended transfer.

Comment: Expiration of device ID is unpredictable.

Conclusion: Quick list consensus on adding device ID.  Recommend GRUU
instead for transfer case.

Caller Prefs

Comment: Enumeration is better

Open Issue: redirection - RFC 3261 proxy merging q-values is broken.
Proposal: include text saying this.

Question: Do we need to mandate this?  Questioner will send a short use =
case
to mailing list.

Open Issue: lost use cases due to changes.

Comment: This is a feature.

Comment: Can be done with multiple requests.

Conclusion: no changes needed.

Open Issues in Use Cases: No discussion.

Comment: Does basing on RFC 2533 provide any value?

Comment: Should RFC 2533 reference just be an informational reference?


SIP Identity - Jon Peterson
-------------------------------------
Topic: AIB

No issues or comments.

Topic: AES and S/MIME

Question: Should we redo S/MIME examples in RFC 3261?

Proposal: Cullen Jennings could do some examples.=20

Comment: Base-64 encoding issue causes interoperability problems. =
(Binary
encoding is better)

Comment: No commercial SIP stacks support S/MIME and TLS.

Comment: There were 2 implementations of S/MIME at last SIPit.


History Info - Mary Barnes
-------------------------------------
Open Issue: Index.  Proposal: Make it mandatory and clarify loose =
routing
behavior.

Open Issue: Internal Retargeting.  Proposal: Include some normative text =
and
examples.

Open Issue: Privacy.  Proposal: Add text.

Comment: Draft needs major clarification on privacy, redirection, =
backwards
compatibility, others. Will discuss on list.

Comment: Include in security section - this header solves a useful =
problem
that a requestor could verify that appropriate proxies have retargeted a
request.

Conclusion: Much more work needed.

Securing SIP Identity Headers - Mary Barnes
--------------------------------------------------------------
(SIPPING draft but discussed here for convenience)

Comment: Question on question not solution.  Do we need to do this?

Comment: This is a type of middle-to-end security problem. We need to =
solve
this problem.

Comment: If we redid Proxy-Auth header, we would use a body instead of a
header.


Parameter Registry - Gonzalo Camarillo
--------------------------------------------------------

Open Issue: Which URI parameters should be registered?

Comment: Do we want p-parameters?

Comment: Lets not make the same mistake twice.

Comment: Want to increase interoperability and avoid collisions.

Comment: To prevent conflicts during Internet-Draft stage, should use =
this
registry.

Comment: C language analogy -- this is like requiring C developers to go
back to Standard C committee and register every variable name they use =
as a
language keyword. The registry could be flooded with requests.  Propose =
that
we should only register URI parameters that are global in nature.

Comment: Have informal non-IANA registry instead (webpage)

Comment: Should do the same thing for parameters (headers and URIs) as
headers.

Comment: Similar to standardization of headers across protocols effort.

Conclusion:   A Hum was taken which supported the creation of an IANA
registry for parameters defined by  RFCs. No consensus on the rest of =
the
issue - more list discussion needed.


Connection Reuse - Rohan Mahy
---------------------------------------------
Open Issue: Clarity on which is original and which is alias.

Open Issue: Security - explaining Mutual TLS and digest

A Hum was taken which supported that people care about the work.

A Hum was taken which supported that this mechanism is reasonable.

Comment: Need to describe how to handle when multiple parties claim the =
same
alias (10.1.1.1).

Conclusion: A Hum was taken which supported that the chairs request a
charter modification to adopt as a WG item.


SIP Security and S/MIME - Cullen Jennings
------------------------------------------------------------
Comment: Mechanism could be used for certificate or raw keys.

Comment: Identity work avoided UAS having to do any PKI operation.  =
Identity
document also only identifies the domain,
not the individual user.

Question: Does this work in both ways?  Answer: Yes, but not exactly.

Many more comments until we ran out of time.

Hum taken on interest of working group in solving this problem.  Strong
interest indicated.

Conclusion: The WG will continue discussion of this topic.


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This list is for NEW development of the core SIP Protocol
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From exim@www1.ietf.org  Tue Jul 22 14:46:47 2003
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Date: Tue, 22 Jul 2003 20:46:28 +0200
Subject: Re: [SIP] Replaces header and REFER
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From: Rohan Mahy <rohan@cisco.com>
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On Wednesday, July 16, 2003, at 03:38 PM, Werkstudent8 SAL wrote:

> From: "Stefan Mecke" <werkstudent8.sal@mch.siemens.de>
>
> Hi all,
>
> I have a question about draft-ietf-sip-replaces-03.txt:
> Section 2 says:
>    ... the Replaces header is
>    frequently used in combination with the REFER [4] method ...

it is frequently used in combination with the REFER method, by 
embedding the Replaces header in the Refer-To URI.   The Replaces 
header NEVER appears in the header section of a REFER request.

> but section 3 contains:
>    ... if
>    a Replaces header field is present in a request other than INVITE,
>    the UAS MUST reject the request ..
>
> I think it makes a lot of sense to allow replaces with REFER; e.g. in 
> the
> call park scenario you could have a web page with all call and make a 
> "click
> to retrieve from park" to have it on your desk phone.
> And that is similar to what's described in
> draft-ietf-sipping-cc-transfer-01.txt, although I have a notion that 
> there
> is some difference but I don't manage to get it out of the draft.

check out draft-ietf-sipping-service-examples

this draft describes several solutions including call park/retrieve.

thanks,
-rohan

> Thanks
>
> _______________________________________________
> 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: "Mary Barnes" <mbarnes@nortelnetworks.com>
To: "'sip@ietf.org'" <sip@ietf.org>
Subject: FW: [Sip] comments on history-info-00
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This email appears to have gotten lost somewhere, so I'm resending (I
apologize if it ends up being a duplicate).

-----Original Message-----
From: Barnes, Mary [NGC:B602:EXCH] 
Sent: Monday, July 21, 2003 1:42 PM
To: 'Jonathan Rosenberg'
Cc: sip@ietf.org
Subject: RE: [Sip] comments on history-info-00


Jonathan,

Some initial responses are embedded below [MB].  I do, however, have some
more general comments that might also address your concerns as follows:

- In general, it seems that alot of your confusion around the fundamental
processing of History-Info is around the fact that we've optimized to only
capturing a single URI, which adds to some of the terminology concerns that
you have. I'm beginning to wonder if it wouldn't be better to go back to
capturing both the retargeted-from and retargeted-to URIs?  The advantage of
this is that it doesn't introduce this "seeding" or "kickstarting" of HI
when a proxy receives a request with no HI, which introduces these special
cases in the fundamental processing. What this also might be highlighting is
the need to bring some of the background text on the solution options from
appendix D into a background section that precedes the normative text for
HI. 

- As I mentioned during the discussion, I fully agree that there is a
general lack of clarity (and as you suggest a need for more precision)
around the fundmental normative processing after discussions with folks that
are planning on implementing this draft. In particular, the processing
around the Index is quite unclear; I think adding the index to all the
examples and adding additional example flows will also help with the concern
here. 

Regards,
Mary H. Barnes
mbarnes@nortelnetworks.com

-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
Sent: Tuesday, July 15, 2003 7:58 PM
To: sip@ietf.org
Subject: [Sip] comments on history-info-00


Mary,

Here are some comments on history-info. I suspect many apply to prior 
versions as well. My apologies if they have been discussed, I have not 
really followed the work closely. Many of my comments are asking for 
much more precision in the document. Generally, I found it to be very 
imprecise, with lots of undefined terminology and vague procedures.

That aside, I have some major issues with the privacy mechanisms, with 
the usage of history-info by redirect servers, with sending a 183 with 
history information, and with the backwards compatibility 
implications. So, here we go:

Firstly, the draft says in many places that applications need to 
document the impact of things. For example, Section 1.1 says:

    Thus, it is highly
    recommended that all applications making use of the request history
    information clearly define the impact of the information not being
    available and specify the processing of such a request.

Does this mean there is some kind of application registry where such 
documentation will exist? 
[MB]: No, I think the use of History-Info is no different than any other
optional SIP header and that application scenarios in the form of call flows
will be needed in Informational (or as has become the trend BCPs) to
effectively document the use of the header. 

Does it mean that applications won't be 
interoperable because they will depend on the presence of these 
headers? 
[MB]: Ditto my previous comment.

The text in 1.1 makes it sound like the information would 
only be present if some local policy prevented its insertion.
[MB] I think you mean "didn't prevent" rather than "prevented" in this
preceeding sentence.  And, yes,local policy is one factor that determines
whether the History-Info is inserted.

What about cases where the proxy doesn't support it? 
[MB]: Then, the information isn't captured for any retargeting by that
proxy, but any previous HI entries should remain, since proxies should not
remove headers. 

Even if it does, your 
draft reads very much like it is now going to require every proxy to 
insert this header. Is that so?
[MB]: No, it's an optional header. However, for systems where this isn't
supported, the functionality of an end application might be limited. 

Second, section 1.3 talks about the use of the Privacy header. There 
are a few problems here. First, the privacy header gets stripped once 
the privacy service finishes execution. Thus, a downstream proxy would 
not be able to know privacy was requested. To a large degree, this is 
because the privacy header in a request is about privacy of the 
caller. Request history is about the current target (i.e., called 
party). So, in fact, the issue is that if the *called* party requests 
privacy (using a privacy header in a response), the history info 
headers should not be propagated backwards in a response. I dont think 
your draft does that.

[MB]: This is similar to several threads of discussion we had around the
requirements about 6 months ago:
http://www1.ietf.org/mail-archive/working-groups/sipping/current/msg03695.ht
ml
which I believe was addressed in a subsequent revision to the requirements
(although, I'll have to dig for the details on that as I don't see that
readily visible in the archives or cached in my email archives at this
time).        

Local policy might also play a role, when it is representing the 
privacy interests of a user that is retargeting. In that case, it was 
unclear to me how the proxy prevents downstream elements from 
inserting history-info? Your draft makes it sound like the privacy 
requirements are met if just this one hop avoids insertion. But seeing 
the contact from downstream retargets will often reveal sensitive 
information about a target further upstream.

[MB]: I agree, I think local policy is this primary mechanism for the
privacy solution for HI and per the email discussion.  As I highlighted in
the meeting, the privacy aspects of the solution are indeed incomplete and
hopefully will be addressed adequately in the next version. 

Section 2.1:

> Targeted-to-URI: the Request URI captured as the Request is 
>         targeted. By capturing a copy of the Request URI in the initial 
>         request, the Retargeted-from-URI is already captured when a 
>         request is retargeted and the Retargeted-to-URI is being 
>         captured.   
>  

I read this many times and could not make sense of it. What does it mean?

[MB] This has to be read in context of the definitions for
Retargeted-from-URI and Retargeted-to-URI in the Definitions section.
Because we're optimizing by capturing only a single URI (the "new" URI in
the request being retargeted) rather than capturing both the "old" and "new"
URIs for each retargeting (and thus duplicating information since the "new"
of the previous HI entry would become the "old" for the next entry). 

Section 2.3.1:

> The UAC SHOULD include the HistInfo option tag in the Supported 
>    header in any request not associated with an established dialog for 
>    which the UAC would like the History-Info in the Response.  In 
>    addition, the UAC should initiate the capturing of the History 
>    Information by capturing the Request-URI as the hi-targeted-to-uri 
>    and initializing the index to 1.  

What if the UAC doesn't support this extension? Can the UAS get 
history info? My understanding is that most, if not all, evnisioned 
applications were for the UAS. This means the UAS can't get 
information unless the caller and many, if not all, proxies along the 
path support it.

[MB] If the UAC doesn't include the option tag, then that means that it
won't get the History-Info in any responses.  Whether the proxy captures
History-Info is local policy; the text is attempting to state that if the
proxy is planning on capturing History-Info then it goes ahead and creates
the "zero-th" entry.  We originally were capturing 2 URIs, the original
request URI and the retargeted-to URI. However, to minimize the overhead and
duplicate data, we made the change in the -01 individual draft to only
capture one URI, so that you basically need to "seed" the chain of URIs to
capture the History, by capturing the original URI at the time of the first
re-targeting that you're intending to capture.  So, the UAS can get the
information even if the caller doesn't want it in responses.  You are
correct that for many applications, it would be necessary for many of the
proxies along the path to support it.  

Section 2.3.2:

> The processing of History-Info by a UAS in a Request depends upon 
>    local policy and specific applications at the UAS which might make 
>    use of the information.  If the HistInfo option tag is received in a 
>    request, the UAS should include any History-Info received in the 
>    request in the subsequent response.     

I think you mean SHOULD in the last sentence. Also, I think you mean 
that "If the request contained the histinfo option tag in the 
Supported header field of the request, the UAS....".

[MB] Correct.

Also, you've capitalized the H and I in HistInfo. Please don't. Option 
tags are not case sensitive.

[MB] Noted.

Section 2.3.3:

The wording about when to capture is really vague. You need to be much 
more precise in your wording of when the proxy is and isnt going to 
capture history information.

[MB]: As I mentioned during my presentation of issues during the WG meeting,
this was the same feedback I had received from some folks that in the
process of implementing this and I do improve this in the next version.
However, this section was intended to be a higher level overview of the
processing with the subsequent sub-sections intended to provide the details
of the normative processing. 

Section 2.3.3.1:

>  If the proxy supports History-Info, the proxy SHOULD add any History-
>    Info collected as it retargets a Request. For retargets that are the 
>    result of an explicit SIP response, the SIP Response Code that 
>    triggered the retargeting MUST be included in the Reason header of 
>    the Targeted-to-URI.

This needs to be more precise. What does it mean "..collected as it 
retargets a request"? How is this information collected? Also, when 
you say "explicit SIP response" - all responses are explicit as far as 
I know. Perhaps you mean when a 3xx response is received? Also, you 
say "Reason header of the Targeted-to-URI". I believe reason is a 
parameter, not a header, in this instance.

and then:
> For retargets as a result of timeouts or 
>    internal events, a Reason header MAY be included in the Reason header 
>    of the Targeted-to-URI. 

what are internal events?

[MB]: This was also something I mentioned at the meeting and has to do with
"internal retargeting" which can be the result of things like selecting the
route in a route list, etc. ******************************************

What value does the reason parameter have in the case of timeout? In 
the case of internal events?

[MB]: It lets an application or end user know why the retargeting occurred.

and then:
>  Additionally, if a request is received that doesn't include a 
>    captured Request URI from the previous entity, the proxy MAY add an 
>    additional entry, effectively capturing the retargeted-from-URI in 
>    the Request.   


What does it mean for a request to "not include a captured 
Request-URI"? You need to define that. What is contained in the 
additional entry?

[MB]: Per my general comment at the beginning of my email response, this has
to do with the current mechanism of only capturing a single URI. 

I may sound like I am being nit-picky, but in my experience 
specifications need to be very precise in what you want people to do. 
I am confused about what is implied here, and I would imagine many 
others would be confused too.

[MB]: Again, as I mentioned during the meeting, this is an open issue with
the draft that I plan to correct on the next version.

and then:

> In order to maintain ordering and accurately reflect the nesting and 
>    retargeting of the request, an index MUST be included along with the 
>    Targeted-to-URI being captured. The basic rule for adding the index 
>    are to read the value from the previous History-Info, if available, 
>    and capture the index.n as the index for the History-Info being 
>    captured, where n would typically be 1 for a forwarded request. Thus, 
>    the level of nesting of the index reflects the number of hops.

define previous.

what does it mean to "capture the index.n"? what is index? what is .n? 
  You say its "typically 1". When is it 1, and when is it not one?

and then:
> For 
>    retargets within a proxy, the proxy MUST maintain the current level 
>    of nesting by incrementing the lowest/last digit of the index for 
>    each instance of retargeting, thus reflecting the number of retargets 
>    within the proxy.

I cannot understand this at all.

and then:
> n index MUST NOT be added 
>    in the scenario whereby the received request had no History-Info 
>    header and the retargeted-from-URI is being captured for 
>    completeness.

what does "for completeness" mean? You need to be precise.

[MB]: Another issue related to the general mechanism of capturing a single
URI that I mentioned at the beginning of my email response. 

> The lack of Reason headers in the captured Request-URIs should be 
>    indicative of the parallel nature of forking (i.e the Request-URIs 
>    are not the result of retargets, but are rather all simultaneous 
>    Targeted-To URIs.)  

should be indicative? Is it, or isnt it?

[MB] Correct. I'll change this from "should be" to "is"

> A proxy that receives a Request with the HistInfo option tag in the 
>    Supported header, and depending upon a local policy supporting the 
>    capture of History-Info, SHOULD return captured History-Info in 
>    subsequent, provisional and final responses to the Request.  A 183 
>    response MAY be sent explicitly for the purposes of conveying 
>    History-Info prior to the final response. 

I think the first sentence means that the proxy does nothing to 
responses - any history-info in those responses is just passed on. But 
I'm not sure. The sentence says "should return", and return seems like 
an active operation to me. 
[MB]: The use of HI does not at all change the messaging, but rather just
adds the captured information to the response in the scenarios where it has
been captured. It is not intended to be an "active" operation.  

The next sentence actually implies that the 
proxy generate a 183 provisional responses. 
[MB]: No, that's not what was intended.  This sentence was added in response
to a concern raised that HI didn't appear to be allowed to be sent in
provisional responses.  

This means that a UA 
making a call with N proxies will get N extra provisional responses, 
where the first has one history-info value, the next 2, the next 3, 
and so on. The first history info is transmitted to the UAC N times. 
Is that really what we want? It seems very expensive for the UAC.
[MB]: Per my response to a previous comment above, HI does not at all change
the fundamental messaging, but rather just adds the captured information to
the response in the scenarios where it has been captured.


> It MAY be advantageous for redirect servers to support the receipt of 
>    History-Info in requests.

You can you assign a protocol strength to being advantageous?

I think you mean "A redirect server MAY use information in the 
History-Info header field in a request to determine the URIs that it 
will retarget to.".

[MB] Agreed; your suggested wording is a more accurate representation of the
intent of the original statement. 

> By receiving it in the request, the 
>    Redirect Server MAY be able to optimize the information it sends in 
>    responses by looking at the already targeted-to-URIs.

and what optimization is that? I think you mean that the redirect 
server shouldnt redirect to somewhere that the request has already 
visited. If so, please say that. However, are we sure about that? 
Redirection to a URI already visited can be appropriate in cases where 
there is a spiral, and not a loop. Unless a redirect server is 
maintaining state, how can it make this determiantionabout whether 
this is a spiral or a loop?

[MB]: You bring up a valid point that should be clarified.  It might be
better to just remove that statement since it really is an implementation
decision and as you suggest such "optimizations" could be problematic in
some scenarios.  (Although, it might be better to highlight the cautions
should someone want to make such an optimization or highlight perhaps that
it's better not to do such an optimization). 



> HistInfo      When used with the Supported header, [RFCXXXX] 
>                  this option tag indicates support 
>                  for the History Information to be  
>                  captured for requests and returned in 
>                  subsequent responses. This tag is not 
>                  used in a Proxy-Require or Requires  
>                  header field since support of  
>                  History-Info is optional.       


Require header field, not Requires.

[MB]: Noted. 


Thanks,
Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com




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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
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Subject: [Sip] When does "Re-INVITE" take effect ?
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Hello,
 
If IP/port in SDP changes, when does "Re-INVITE" take effect ?
 
from Sip-Implementors mail archive
http://lists.cs.columbia.edu/pipermail/sip-implementors/2001-May/001211.html
<http://lists.cs.columbia.edu/pipermail/sip-implementors/2001-May/001211.htm
l> 
 
from RFC3312
Integration of Resource Management and Session Initiation Protocol (SIP) 
13. 1 End-to-End Status Type
http://www.zvon.org/tmRFC/RFC3312/Output/chapter13.html
<http://www.zvon.org/tmRFC/RFC3312/Output/chapter13.html> 
 
from RFC3264 
An Offer/Answer Model Session Description Protocol
8.3.1 Modifying Address, Port or Transport
http://www.zvon.org/tmRFC/RFC3264/Output/chapter8.html
<http://www.zvon.org/tmRFC/RFC3264/Output/chapter8.html> 
 
        Alice                Proxy 2                Bob 

           |      Both Way RTP Media Established     | 

           |<=======================================>| 

           |                                         | 

           |           Bob changes IP address        | 

           |                                         | 

           |                 F9 INVITE               | 

           |<----------------------------------------| 

           |                F10 200 OK               | 

           |---------------------------------------->|   

           |                 F11  ACK                | 

           |<----------------------------------------| 

           |         New RTP Media Stream            | 

           |<=======================================>| 

 

1) In above flows, 

Alice may continue sending media to Bob's old address during "Re-INVITE".

Alice may start sending media to Bob's new address 

before sending 200 OK, if "Re-INVITE" is accepted by Alice.

Bob may continue sending media to Alice during "Re-INVITE" ?

Am I right ?

 

2) If RTCP timeout occurs on Alice and there is no active media session with
Bob, 

does Alice terminate SIP session with Bob ?

 

3) If Bob sends RTCP with new SSRC and new address before "Re-INVITE" is
accepted by Alice, 

is there any problem in that ?

 

Thanks in advance.

 

Brad Park


------_=_NextPart_001_01C3511A.1245A6E0
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charset=3Dus-ascii">
<TITLE>&#47700;&#49884;&#51648;</TITLE>

<META content=3D"MSHTML 6.00.2800.1106" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3D&#44404;&#47548; size=3D2>
<DIV><SPAN class=3D687433208-23072003><FONT =
face=3D&#44404;&#47548;&#52404;=20
size=3D2>Hello,</FONT></SPAN></DIV>
<DIV><SPAN class=3D687433208-23072003><FONT =
face=3D&#44404;&#47548;&#52404;=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D687433208-23072003><FONT =
face=3D&#44404;&#47548;&#52404; size=3D2>If IP/port in=20
SDP&nbsp;changes, when does "Re-INVITE" take effect =
?</FONT></SPAN></DIV>
<DIV><SPAN class=3D687433208-23072003><FONT =
face=3D&#44404;&#47548;&#52404;=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D687433208-23072003><FONT =
face=3D&#44404;&#47548;&#52404; size=3D2>from Sip-Implementors=20
mail archive</FONT></SPAN></DIV>
<DIV><SPAN class=3D687433208-23072003><FONT =
face=3D&#44404;&#47548;&#52404; size=3D2><A=20
href=3D"http://lists.cs.columbia.edu/pipermail/sip-implementors/2001-May=
/001211.html">http://lists.cs.columbia.edu/pipermail/sip-implementors/20=
01-May/001211.html</A></FONT></SPAN></DIV>
<DIV><SPAN class=3D687433208-23072003><FONT =
face=3D&#44404;&#47548;&#52404;=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D687433208-23072003><FONT =
face=3D&#44404;&#47548;&#52404; size=3D2>from=20
RFC3312</FONT></SPAN></DIV>
<DIV><SPAN class=3D687433208-23072003><FONT =
face=3D&#44404;&#47548;&#52404;><FONT size=3D2><FONT=20
face=3D&#44404;&#47548;>Integration of Resource Management and Session =
Initiation Protocol=20
(SIP)</FONT> </FONT></DIV>
<DIV><SPAN class=3D687433208-23072003><FONT face=3D&#44404;&#47548; =
size=3D2>13. 1 End-to-End Status=20
Type</FONT></SPAN></FONT></SPAN></DIV>
<DIV><SPAN class=3D687433208-23072003><FONT =
face=3D&#44404;&#47548;&#52404; size=3D2><A=20
href=3D"http://www.zvon.org/tmRFC/RFC3312/Output/chapter13.html">http://=
www.zvon.org/tmRFC/RFC3312/Output/chapter13.html</A></FONT></SPAN></DIV>=

<DIV><SPAN class=3D687433208-23072003><FONT =
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D687433208-23072003><FONT size=3D2>from RFC3264=20
</FONT></SPAN></DIV>
<DIV><SPAN class=3D687433208-23072003><FONT size=3D2><SPAN lang=3DEN-US =

style=3D"FONT-SIZE: 10pt; FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;; mso-bidi-font-size: =
12.0pt; mso-font-kerning: 1.0pt; mso-ansi-language: EN-US; =
mso-fareast-language: KO; mso-bidi-language: AR-SA">An=20
Offer/Answer Model Session Description =
Protocol</SPAN></FONT></SPAN></DIV>
<DIV><SPAN class=3D687433208-23072003><FONT size=3D2><SPAN lang=3DEN-US =

style=3D"FONT-SIZE: 10pt; FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;; mso-bidi-font-size: =
12.0pt; mso-font-kerning: 1.0pt; mso-ansi-language: EN-US; =
mso-fareast-language: KO; mso-bidi-language: AR-SA">8.3.1=20
Modifying Address, Port or Transport</SPAN></FONT></SPAN></DIV>
<DIV><SPAN class=3D687433208-23072003><FONT size=3D2><A=20
href=3D"http://www.zvon.org/tmRFC/RFC3264/Output/chapter8.html">http://w=
ww.zvon.org/tmRFC/RFC3264/Output/chapter8.html</A></FONT></SPAN></DIV>
<DIV><SPAN class=3D687433208-23072003><FONT =
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D687433208-23072003>
<P class=3DMsoPlainText=20
style=3D"MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: 11pt; mso-line-height-rule: =
exactly"><SPAN=20
lang=3DEN-US style=3D"FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;"><FONT=20
size=3D2><SPAN=20
style=3D"mso-spacerun: yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
</SPAN>Alice<SPAN=20
style=3D"mso-spacerun: =
yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;=20
</SPAN>Proxy 2<SPAN=20
style=3D"mso-spacerun: =
yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;=20
</SPAN>Bob <?xml:namespace prefix =3D o ns =3D=20
"urn:schemas-microsoft-com:office:office" =
/><o:p></o:p></FONT></SPAN></P>
<P class=3DMsoPlainText=20
style=3D"MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: 11pt; mso-line-height-rule: =
exactly"><SPAN=20
lang=3DEN-US style=3D"FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;"><FONT=20
size=3D2><SPAN=20
style=3D"mso-spacerun: =
yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
</SPAN>|<SPAN style=3D"mso-spacerun: =
yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
</SPAN>Both Way RTP Media Established<SPAN=20
style=3D"mso-spacerun: yes">&nbsp;&nbsp;&nbsp;&nbsp; </SPAN>|=20
<o:p></o:p></FONT></SPAN></P>
<P class=3DMsoPlainText=20
style=3D"MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: 11pt; mso-line-height-rule: =
exactly"><SPAN=20
lang=3DEN-US style=3D"FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;"><FONT=20
size=3D2><SPAN=20
style=3D"mso-spacerun: =
yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
</SPAN>|&lt;=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D&gt;|=20
<o:p></o:p></FONT></SPAN></P>
<P class=3DMsoPlainText=20
style=3D"MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: 11pt; mso-line-height-rule: =
exactly"><SPAN=20
lang=3DEN-US style=3D"FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;"><FONT=20
size=3D2><SPAN=20
style=3D"mso-spacerun: =
yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
</SPAN>|<SPAN=20
style=3D"mso-spacerun: =
yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
</SPAN>| <o:p></o:p></FONT></SPAN></P>
<P class=3DMsoPlainText=20
style=3D"MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: 11pt; mso-line-height-rule: =
exactly"><SPAN=20
lang=3DEN-US style=3D"FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;"><FONT=20
size=3D2><SPAN=20
style=3D"mso-spacerun: =
yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
</SPAN>|<SPAN=20
style=3D"mso-spacerun: =
yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
</SPAN>Bob changes IP address<SPAN=20
style=3D"mso-spacerun: yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</SPAN>|=20
<o:p></o:p></FONT></SPAN></P>
<P class=3DMsoPlainText=20
style=3D"MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: 11pt; mso-line-height-rule: =
exactly"><SPAN=20
lang=3DEN-US style=3D"FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;"><FONT=20
size=3D2><SPAN=20
style=3D"mso-spacerun: =
yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
</SPAN>|<SPAN=20
style=3D"mso-spacerun: =
yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
</SPAN>| <o:p></o:p></FONT></SPAN></P>
<P class=3DMsoPlainText=20
style=3D"MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: 11pt; mso-line-height-rule: =
exactly"><SPAN=20
lang=3DEN-US style=3D"FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;"><FONT=20
size=3D2><SPAN=20
style=3D"mso-spacerun: =
yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
</SPAN>|<SPAN=20
style=3D"mso-spacerun: =
yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
</SPAN>F9 INVITE<SPAN=20
style=3D"mso-spacerun: =
yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;=20
</SPAN>| <o:p></o:p></FONT></SPAN></P>
<P class=3DMsoPlainText=20
style=3D"MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: 11pt; mso-line-height-rule: =
exactly"><SPAN=20
lang=3DEN-US style=3D"FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;"><FONT=20
size=3D2><SPAN=20
style=3D"mso-spacerun: =
yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
</SPAN>|&lt;----------------------------------------|=20
<o:p></o:p></FONT></SPAN></P>
<P class=3DMsoPlainText=20
style=3D"MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: 11pt; mso-line-height-rule: =
exactly"><SPAN=20
lang=3DEN-US style=3D"FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;"><FONT=20
size=3D2><SPAN=20
style=3D"mso-spacerun: =
yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
</SPAN>|<SPAN=20
style=3D"mso-spacerun: =
yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;=20
</SPAN>F10 200 OK<SPAN=20
style=3D"mso-spacerun: =
yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;=20
</SPAN>| <o:p></o:p></FONT></SPAN></P>
<P class=3DMsoPlainText=20
style=3D"MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: 11pt; mso-line-height-rule: =
exactly"><SPAN=20
lang=3DEN-US style=3D"FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;"><FONT=20
size=3D2><SPAN=20
style=3D"mso-spacerun: =
yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
</SPAN>|----------------------------------------&gt;|<SPAN=20
style=3D"mso-spacerun: yes">&nbsp;&nbsp; =
</SPAN><o:p></o:p></FONT></SPAN></P>
<P class=3DMsoPlainText=20
style=3D"MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: 11pt; mso-line-height-rule: =
exactly"><SPAN=20
lang=3DEN-US style=3D"FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;"><FONT=20
size=3D2><SPAN=20
style=3D"mso-spacerun: =
yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
</SPAN>|<SPAN=20
style=3D"mso-spacerun: =
yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
</SPAN>F11<SPAN style=3D"mso-spacerun: yes">&nbsp; </SPAN>ACK<SPAN=20
style=3D"mso-spacerun: =
yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;=20
</SPAN>| <o:p></o:p></FONT></SPAN></P>
<P class=3DMsoPlainText=20
style=3D"MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: 11pt; mso-line-height-rule: =
exactly"><SPAN=20
lang=3DEN-US style=3D"FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;"><FONT=20
size=3D2><SPAN=20
style=3D"mso-spacerun: =
yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
</SPAN>|&lt;----------------------------------------|=20
<o:p></o:p></FONT></SPAN></P>
<P class=3DMsoPlainText=20
style=3D"MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: 11pt; mso-line-height-rule: =
exactly"><SPAN=20
lang=3DEN-US style=3D"FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;"><FONT=20
size=3D2><SPAN=20
style=3D"mso-spacerun: =
yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
</SPAN>|<SPAN=20
style=3D"mso-spacerun: =
yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
</SPAN>New RTP Media Stream<SPAN=20
style=3D"mso-spacerun: =
yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =

</SPAN>| <o:p></o:p></FONT></SPAN></P>
<P class=3DMsoPlainText=20
style=3D"MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: 11pt; mso-line-height-rule: =
exactly"><SPAN=20
lang=3DEN-US style=3D"FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;"><FONT=20
size=3D2><SPAN=20
style=3D"mso-spacerun: =
yes">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
</SPAN>|&lt;=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D&gt;|=20
<o:p></o:p></FONT></SPAN></P>
<P class=3DMsoPlainText=20
style=3D"MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: 11pt; mso-line-height-rule: =
exactly"><SPAN=20
lang=3DEN-US style=3D"FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;"><FONT=20
size=3D2></FONT></SPAN>&nbsp;</P>
<P class=3DMsoPlainText=20
style=3D"MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: 11pt; mso-line-height-rule: =
exactly"><SPAN=20
lang=3DEN-US style=3D"FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;"><SPAN=20
class=3D687433208-23072003><FONT size=3D2>1) In above flows,=20
</FONT></SPAN></SPAN></P>
<P class=3DMsoPlainText=20
style=3D"MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: 11pt; mso-line-height-rule: =
exactly"><SPAN=20
lang=3DEN-US style=3D"FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;"><FONT=20
size=3D2><SPAN class=3D687433208-23072003>Alice may continue sending =
media to Bob's=20
old address during "Re-INVITE".</SPAN></FONT></SPAN></P>
<P class=3DMsoPlainText=20
style=3D"MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: 11pt; mso-line-height-rule: =
exactly"><SPAN=20
lang=3DEN-US style=3D"FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;"><FONT=20
size=3D2><SPAN class=3D687433208-23072003>Alice may start sending media =
to Bob's new=20
address </SPAN></FONT></SPAN></P>
<P class=3DMsoPlainText=20
style=3D"MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: 11pt; mso-line-height-rule: =
exactly"><SPAN=20
lang=3DEN-US style=3D"FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;"><FONT=20
size=3D2><SPAN class=3D687433208-23072003>before&nbsp;sending 200 OK, =
if "Re-INVITE"=20
is accepted by Alice.</SPAN></FONT></SPAN></P>
<P class=3DMsoPlainText=20
style=3D"MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: 11pt; mso-line-height-rule: =
exactly"><SPAN=20
lang=3DEN-US style=3D"FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;"><FONT=20
size=3D2><SPAN class=3D687433208-23072003>Bob may continue sending =
media to Alice=20
during "Re-INVITE" ?</SPAN></FONT></SPAN></P>
<P class=3DMsoPlainText=20
style=3D"MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: 11pt; mso-line-height-rule: =
exactly"><SPAN=20
lang=3DEN-US style=3D"FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;"><FONT=20
size=3D2><SPAN class=3D687433208-23072003>Am I right =
?</SPAN></FONT></SPAN></P>
<P class=3DMsoPlainText=20
style=3D"MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: 11pt; mso-line-height-rule: =
exactly"><SPAN=20
lang=3DEN-US style=3D"FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;"><FONT=20
size=3D2><SPAN =
class=3D687433208-23072003></SPAN></FONT></SPAN>&nbsp;</P>
<P class=3DMsoPlainText=20
style=3D"MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: 11pt; mso-line-height-rule: =
exactly"><SPAN=20
lang=3DEN-US style=3D"FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;"><FONT=20
size=3D2><SPAN class=3D687433208-23072003>2) If RTCP timeout =
occurs&nbsp;on Alice=20
and there is no active media session with Bob, =
</SPAN></FONT></SPAN></P>
<P class=3DMsoPlainText=20
style=3D"MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: 11pt; mso-line-height-rule: =
exactly"><SPAN=20
lang=3DEN-US style=3D"FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;"><FONT=20
size=3D2><SPAN class=3D687433208-23072003>does Alice terminate SIP =
session with Bob=20
?</SPAN></FONT></SPAN></P>
<P class=3DMsoPlainText=20
style=3D"MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: 11pt; mso-line-height-rule: =
exactly"><SPAN=20
lang=3DEN-US style=3D"FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;"><FONT=20
size=3D2><SPAN =
class=3D687433208-23072003></SPAN></FONT></SPAN>&nbsp;</P>
<P class=3DMsoPlainText=20
style=3D"MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: 11pt; mso-line-height-rule: =
exactly"><SPAN=20
lang=3DEN-US style=3D"FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;"><FONT=20
size=3D2><SPAN class=3D687433208-23072003>3) If Bob sends RTCP with new =
SSRC and new=20
address before "Re-INVITE" is accepted by Alice, =
</SPAN></FONT></SPAN></P>
<P class=3DMsoPlainText=20
style=3D"MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: 11pt; mso-line-height-rule: =
exactly"><SPAN=20
lang=3DEN-US style=3D"FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;"><FONT=20
size=3D2><SPAN class=3D687433208-23072003>is there any problem in that=20
?</SPAN></FONT></SPAN></P>
<P class=3DMsoPlainText=20
style=3D"MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: 11pt; mso-line-height-rule: =
exactly"><SPAN=20
lang=3DEN-US style=3D"FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;"><FONT=20
size=3D2><SPAN =
class=3D687433208-23072003></SPAN></FONT></SPAN>&nbsp;</P>
<P class=3DMsoPlainText=20
style=3D"MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: 11pt; mso-line-height-rule: =
exactly"><SPAN=20
lang=3DEN-US style=3D"FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;"><FONT=20
size=3D2><SPAN class=3D687433208-23072003>Thanks in=20
advance.</SPAN></FONT></SPAN></P>
<P class=3DMsoPlainText=20
style=3D"MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: 11pt; mso-line-height-rule: =
exactly"><SPAN=20
lang=3DEN-US style=3D"FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;"><FONT=20
size=3D2><SPAN =
class=3D687433208-23072003></SPAN></FONT></SPAN>&nbsp;</P>
<P class=3DMsoPlainText=20
style=3D"MARGIN: 0cm 0cm 0pt; LINE-HEIGHT: 11pt; mso-line-height-rule: =
exactly"><SPAN=20
lang=3DEN-US style=3D"FONT-FAMILY: &#44404;&#47548;&#52404;; =
mso-bidi-font-family: &#44404;&#47548;&#52404;"><FONT=20
size=3D2><SPAN=20
class=3D687433208-23072003>Brad&nbsp;Park</SPAN></FONT></SPAN></P></SPAN=
></DIV></FONT></DIV></BODY></HTML>

------_=_NextPart_001_01C3511A.1245A6E0--

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From exim@www1.ietf.org  Wed Jul 23 10:08:35 2003
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Subject: [Sip] non-INVITE transaction issues
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Hi Folks,

Last week in Vienna we skipped over a presentation from Robert Sparks 
about non-INVITE transactions, as he was very ill at the time.  I'd 
like to start up a thread about the non-INVITE transaction draft that 
he wrote and the issues discussed there.  This is important work, and 
some documents will depend on our solutions (guidelines, and new 
methods). Please review the document, and provide comments.  Also, 
Robert may provide the presentation he was going to give which should 
offer some guidance.

many thanks,
-rohan
co-chair SIP and SIPPING


(Note: I will send my own comments as an individual as a separate 
response to this message.)




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From exim@www1.ietf.org  Wed Jul 23 12:11:47 2003
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Thread-Topic: Is there any standard SIP/SDP for communicating echo cancellation ?
Thread-Index: AcNRNCPkW4h+cxIlSRGXv/w4xvJGIg==
From: "Hashmi, Mohamed Hadi" <Hadi.Hashmi@Level3.com>
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Subject: [Sip] Is there any standard SIP/SDP for communicating echo cancellation ?
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For transmitting FAX or Modem call over VoIP  ( RTP- preferably G.711) =
in PSTN-SIP-PSTN call it is required that echo cancellation be turned =
off at both end Gateways. There are specific standards for FAX (T.38) =
and MoIP (V.150.1) to deal with these issues, but not every vendor has =
implemented those standards in their Media Gateways.=20

In such a situation  G.711 with echo cancellation turned off at both end =
will help transmitting such calls like FAX or  MoIP. Gateway control =
protocols such as MGCP,  Megaco,  IPDC  etc do have specific tags to =
handle this case. But don't know how this information can be =
communicated between two Gateway Controllers using SIP.  Is anybody =
aware of any standard SDP  for such a functionality?  Although an SDP =
can be custom designed for this purpose but this wouldn't help =
interoperation between two different vendors.=20

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Subject: Re: [Sip] Is there any standard SIP/SDP for communicating echo cancellation ?
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Hi,

It would probably help if you took a look at 
draft-ietf-sipping-realtimefax-01.txt and its references.  If you still 
have questions, please post your question to the MMUSIC WG mailing list.

many thanks,
-rohan
co-chair SIP and SIPPING


On Wednesday, July 23, 2003, at 06:04 PM, Hashmi, Mohamed Hadi wrote:

> For transmitting FAX or Modem call over VoIP  ( RTP- preferably G.711) 
> in PSTN-SIP-PSTN call it is required that echo cancellation be turned 
> off at both end Gateways. There are specific standards for FAX (T.38) 
> and MoIP (V.150.1) to deal with these issues, but not every vendor has 
> implemented those standards in their Media Gateways.
>
> In such a situation  G.711 with echo cancellation turned off at both 
> end will help transmitting such calls like FAX or  MoIP. Gateway 
> control protocols such as MGCP,  Megaco,  IPDC  etc do have specific 
> tags to handle this case. But don't know how this information can be 
> communicated between two Gateway Controllers using SIP.  Is anybody 
> aware of any standard SDP  for such a functionality?  Although an SDP 
> can be custom designed for this purpose but this wouldn't help 
> interoperation between two different vendors.
>
> _______________________________________________
> 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
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From: Adam Roach <adam@dynamicsoft.com>
To: "'Rohan Mahy'" <rohan@cisco.com>, sip@ietf.org
Cc: Robert Sparks <rsparks@dynamicsoft.com>
Subject: RE: [Sip] non-INVITE transaction issues
Date: Wed, 23 Jul 2003 17:29:04 -0500
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I think it would be useful for anyone interested in this
topic to review the minutes from the ad-hoc meeting we
had on this topic in San Francisco. 

Please see:

<http://www1.ietf.org/mail-archive/working-groups/sip/current/msg07845.html>

/a

> -----Original Message-----
> From: Rohan Mahy [mailto:rohan@cisco.com]
> Sent: Wednesday, July 23, 2003 9:08
> To: sip@ietf.org
> Cc: 'Robert Sparks'
> Subject: [Sip] non-INVITE transaction issues
> 
> 
> Hi Folks,
> 
> Last week in Vienna we skipped over a presentation from Robert Sparks 
> about non-INVITE transactions, as he was very ill at the time.  I'd 
> like to start up a thread about the non-INVITE transaction draft that 
> he wrote and the issues discussed there.  This is important work, and 
> some documents will depend on our solutions (guidelines, and new 
> methods). Please review the document, and provide comments.  Also, 
> Robert may provide the presentation he was going to give which should 
> offer some guidance.
> 
> many thanks,
> -rohan
> co-chair SIP and SIPPING
> 
> 
> (Note: I will send my own comments as an individual as a separate 
> response to this message.)
> 
> 
> 
> 
> _______________________________________________
> 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: "Mary Barnes" <mbarnes@nortelnetworks.com>
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Subject: RE: [Sip] comments on history-info-00
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Jonathan,

Some initial responses are embedded below [MB].  I do, however, have some
more general comments that might also address your concerns as follows:

- In general, it seems that alot of your confusion around the fundamental
processing of History-Info is around the fact that we've optimized to only
capturing a single URI, which adds to some of the terminology concerns that
you have. I'm beginning to wonder if it wouldn't be better to go back to
capturing both the retargeted-from and retargeted-to URIs?  The advantage of
this is that it doesn't introduce this "seeding" or "kickstarting" of HI
when a proxy receives a request with no HI, which introduces these special
cases in the fundamental processing. What this also might be highlighting is
the need to bring some of the background text on the solution options from
appendix D into a background section that precedes the normative text for
HI. 

- As I mentioned during the discussion, I fully agree that there is a
general lack of clarity (and as you suggest a need for more precision)
around the fundmental normative processing after discussions with folks that
are planning on implementing this draft. In particular, the processing
around the Index is quite unclear; I think adding the index to all the
examples and adding additional example flows will also help with the concern
here. 

Regards,
Mary H. Barnes
mbarnes@nortelnetworks.com

-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
Sent: Tuesday, July 15, 2003 7:58 PM
To: sip@ietf.org
Subject: [Sip] comments on history-info-00


Mary,

Here are some comments on history-info. I suspect many apply to prior 
versions as well. My apologies if they have been discussed, I have not 
really followed the work closely. Many of my comments are asking for 
much more precision in the document. Generally, I found it to be very 
imprecise, with lots of undefined terminology and vague procedures.

That aside, I have some major issues with the privacy mechanisms, with 
the usage of history-info by redirect servers, with sending a 183 with 
history information, and with the backwards compatibility 
implications. So, here we go:

Firstly, the draft says in many places that applications need to 
document the impact of things. For example, Section 1.1 says:

    Thus, it is highly
    recommended that all applications making use of the request history
    information clearly define the impact of the information not being
    available and specify the processing of such a request.

Does this mean there is some kind of application registry where such 
documentation will exist? 
[MB]: No, I think the use of History-Info is no different than any other
optional SIP header and that application scenarios in the form of call flows
will be needed in Informational (or as has become the trend BCPs) to
effectively document the use of the header. 

Does it mean that applications won't be 
interoperable because they will depend on the presence of these 
headers? 
[MB]: Ditto my previous comment.

The text in 1.1 makes it sound like the information would 
only be present if some local policy prevented its insertion.
[MB] I think you mean "didn't prevent" rather than "prevented" in this
preceeding sentence.  And, yes,local policy is one factor that determines
whether the History-Info is inserted.

What about cases where the proxy doesn't support it? 
[MB]: Then, the information isn't captured for any retargeting by that
proxy, but any previous HI entries should remain, since proxies should not
remove headers. 

Even if it does, your 
draft reads very much like it is now going to require every proxy to 
insert this header. Is that so?
[MB]: No, it's an optional header. However, for systems where this isn't
supported, the functionality of an end application might be limited. 

Second, section 1.3 talks about the use of the Privacy header. There 
are a few problems here. First, the privacy header gets stripped once 
the privacy service finishes execution. Thus, a downstream proxy would 
not be able to know privacy was requested. To a large degree, this is 
because the privacy header in a request is about privacy of the 
caller. Request history is about the current target (i.e., called 
party). So, in fact, the issue is that if the *called* party requests 
privacy (using a privacy header in a response), the history info 
headers should not be propagated backwards in a response. I dont think 
your draft does that.

[MB]: This is similar to several threads of discussion we had around the
requirements about 6 months ago:
http://www1.ietf.org/mail-archive/working-groups/sipping/current/msg03695.ht
ml
which I believe was addressed in a subsequent revision to the requirements
(although, I'll have to dig for the details on that as I don't see that
readily visible in the archives or cached in my email archives at this
time).        

Local policy might also play a role, when it is representing the 
privacy interests of a user that is retargeting. In that case, it was 
unclear to me how the proxy prevents downstream elements from 
inserting history-info? Your draft makes it sound like the privacy 
requirements are met if just this one hop avoids insertion. But seeing 
the contact from downstream retargets will often reveal sensitive 
information about a target further upstream.

[MB]: I agree, I think local policy is this primary mechanism for the
privacy solution for HI and per the email discussion.  As I highlighted in
the meeting, the privacy aspects of the solution are indeed incomplete and
hopefully will be addressed adequately in the next version. 

Section 2.1:

> Targeted-to-URI: the Request URI captured as the Request is 
>         targeted. By capturing a copy of the Request URI in the initial 
>         request, the Retargeted-from-URI is already captured when a 
>         request is retargeted and the Retargeted-to-URI is being 
>         captured.   
>  

I read this many times and could not make sense of it. What does it mean?

[MB] This has to be read in context of the definitions for
Retargeted-from-URI and Retargeted-to-URI in the Definitions section.
Because we're optimizing by capturing only a single URI (the "new" URI in
the request being retargeted) rather than capturing both the "old" and "new"
URIs for each retargeting (and thus duplicating information since the "new"
of the previous HI entry would become the "old" for the next entry). 

Section 2.3.1:

> The UAC SHOULD include the HistInfo option tag in the Supported 
>    header in any request not associated with an established dialog for 
>    which the UAC would like the History-Info in the Response.  In 
>    addition, the UAC should initiate the capturing of the History 
>    Information by capturing the Request-URI as the hi-targeted-to-uri 
>    and initializing the index to 1.  

What if the UAC doesn't support this extension? Can the UAS get 
history info? My understanding is that most, if not all, evnisioned 
applications were for the UAS. This means the UAS can't get 
information unless the caller and many, if not all, proxies along the 
path support it.

[MB] If the UAC doesn't include the option tag, then that means that it
won't get the History-Info in any responses.  Whether the proxy captures
History-Info is local policy; the text is attempting to state that if the
proxy is planning on capturing History-Info then it goes ahead and creates
the "zero-th" entry.  We originally were capturing 2 URIs, the original
request URI and the retargeted-to URI. However, to minimize the overhead and
duplicate data, we made the change in the -01 individual draft to only
capture one URI, so that you basically need to "seed" the chain of URIs to
capture the History, by capturing the original URI at the time of the first
re-targeting that you're intending to capture.  So, the UAS can get the
information even if the caller doesn't want it in responses.  You are
correct that for many applications, it would be necessary for many of the
proxies along the path to support it.  

Section 2.3.2:

> The processing of History-Info by a UAS in a Request depends upon 
>    local policy and specific applications at the UAS which might make 
>    use of the information.  If the HistInfo option tag is received in a 
>    request, the UAS should include any History-Info received in the 
>    request in the subsequent response.     

I think you mean SHOULD in the last sentence. Also, I think you mean 
that "If the request contained the histinfo option tag in the 
Supported header field of the request, the UAS....".

[MB] Correct.

Also, you've capitalized the H and I in HistInfo. Please don't. Option 
tags are not case sensitive.

[MB] Noted.

Section 2.3.3:

The wording about when to capture is really vague. You need to be much 
more precise in your wording of when the proxy is and isnt going to 
capture history information.

[MB]: As I mentioned during my presentation of issues during the WG meeting,
this was the same feedback I had received from some folks that in the
process of implementing this and I do improve this in the next version.
However, this section was intended to be a higher level overview of the
processing with the subsequent sub-sections intended to provide the details
of the normative processing. 

Section 2.3.3.1:

>  If the proxy supports History-Info, the proxy SHOULD add any History-
>    Info collected as it retargets a Request. For retargets that are the 
>    result of an explicit SIP response, the SIP Response Code that 
>    triggered the retargeting MUST be included in the Reason header of 
>    the Targeted-to-URI.

This needs to be more precise. What does it mean "..collected as it 
retargets a request"? How is this information collected? Also, when 
you say "explicit SIP response" - all responses are explicit as far as 
I know. Perhaps you mean when a 3xx response is received? Also, you 
say "Reason header of the Targeted-to-URI". I believe reason is a 
parameter, not a header, in this instance.

and then:
> For retargets as a result of timeouts or 
>    internal events, a Reason header MAY be included in the Reason header 
>    of the Targeted-to-URI. 

what are internal events?

[MB]: This was also something I mentioned at the meeting and has to do with
"internal retargeting" which can be the result of things like selecting the
route in a route list, etc. ******************************************

What value does the reason parameter have in the case of timeout? In 
the case of internal events?

[MB]: It lets an application or end user know why the retargeting occurred.

and then:
>  Additionally, if a request is received that doesn't include a 
>    captured Request URI from the previous entity, the proxy MAY add an 
>    additional entry, effectively capturing the retargeted-from-URI in 
>    the Request.   


What does it mean for a request to "not include a captured 
Request-URI"? You need to define that. What is contained in the 
additional entry?

[MB]: Per my general comment at the beginning of my email response, this has
to do with the current mechanism of only capturing a single URI. 

I may sound like I am being nit-picky, but in my experience 
specifications need to be very precise in what you want people to do. 
I am confused about what is implied here, and I would imagine many 
others would be confused too.

[MB]: Again, as I mentioned during the meeting, this is an open issue with
the draft that I plan to correct on the next version.

and then:

> In order to maintain ordering and accurately reflect the nesting and 
>    retargeting of the request, an index MUST be included along with the 
>    Targeted-to-URI being captured. The basic rule for adding the index 
>    are to read the value from the previous History-Info, if available, 
>    and capture the index.n as the index for the History-Info being 
>    captured, where n would typically be 1 for a forwarded request. Thus, 
>    the level of nesting of the index reflects the number of hops.

define previous.

what does it mean to "capture the index.n"? what is index? what is .n? 
  You say its "typically 1". When is it 1, and when is it not one?

and then:
> For 
>    retargets within a proxy, the proxy MUST maintain the current level 
>    of nesting by incrementing the lowest/last digit of the index for 
>    each instance of retargeting, thus reflecting the number of retargets 
>    within the proxy.

I cannot understand this at all.

and then:
> n index MUST NOT be added 
>    in the scenario whereby the received request had no History-Info 
>    header and the retargeted-from-URI is being captured for 
>    completeness.

what does "for completeness" mean? You need to be precise.

[MB]: Another issue related to the general mechanism of capturing a single
URI that I mentioned at the beginning of my email response. 

> The lack of Reason headers in the captured Request-URIs should be 
>    indicative of the parallel nature of forking (i.e the Request-URIs 
>    are not the result of retargets, but are rather all simultaneous 
>    Targeted-To URIs.)  

should be indicative? Is it, or isnt it?

[MB] Correct. I'll change this from "should be" to "is"

> A proxy that receives a Request with the HistInfo option tag in the 
>    Supported header, and depending upon a local policy supporting the 
>    capture of History-Info, SHOULD return captured History-Info in 
>    subsequent, provisional and final responses to the Request.  A 183 
>    response MAY be sent explicitly for the purposes of conveying 
>    History-Info prior to the final response. 

I think the first sentence means that the proxy does nothing to 
responses - any history-info in those responses is just passed on. But 
I'm not sure. The sentence says "should return", and return seems like 
an active operation to me. 
[MB]: The use of HI does not at all change the messaging, but rather just
adds the captured information to the response in the scenarios where it has
been captured. It is not intended to be an "active" operation.  

The next sentence actually implies that the 
proxy generate a 183 provisional responses. 
[MB]: No, that's not what was intended.  This sentence was added in response
to a concern raised that HI didn't appear to be allowed to be sent in
provisional responses.  

This means that a UA 
making a call with N proxies will get N extra provisional responses, 
where the first has one history-info value, the next 2, the next 3, 
and so on. The first history info is transmitted to the UAC N times. 
Is that really what we want? It seems very expensive for the UAC.
[MB]: Per my response to a previous comment above, HI does not at all change
the fundamental messaging, but rather just adds the captured information to
the response in the scenarios where it has been captured.


> It MAY be advantageous for redirect servers to support the receipt of 
>    History-Info in requests.

You can you assign a protocol strength to being advantageous?

I think you mean "A redirect server MAY use information in the 
History-Info header field in a request to determine the URIs that it 
will retarget to.".

[MB] Agreed; your suggested wording is a more accurate representation of the
intent of the original statement. 

> By receiving it in the request, the 
>    Redirect Server MAY be able to optimize the information it sends in 
>    responses by looking at the already targeted-to-URIs.

and what optimization is that? I think you mean that the redirect 
server shouldnt redirect to somewhere that the request has already 
visited. If so, please say that. However, are we sure about that? 
Redirection to a URI already visited can be appropriate in cases where 
there is a spiral, and not a loop. Unless a redirect server is 
maintaining state, how can it make this determiantionabout whether 
this is a spiral or a loop?

[MB]: You bring up a valid point that should be clarified.  It might be
better to just remove that statement since it really is an implementation
decision and as you suggest such "optimizations" could be problematic in
some scenarios.  (Although, it might be better to highlight the cautions
should someone want to make such an optimization or highlight perhaps that
it's better not to do such an optimization). 



> HistInfo      When used with the Supported header, [RFCXXXX] 
>                  this option tag indicates support 
>                  for the History Information to be  
>                  captured for requests and returned in 
>                  subsequent responses. This tag is not 
>                  used in a Proxy-Require or Requires  
>                  header field since support of  
>                  History-Info is optional.       


Require header field, not Requires.

[MB]: Noted. 


Thanks,
Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com




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<TITLE>RE: [Sip] comments on history-info-00</TITLE>
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<P><FONT SIZE=3D2>Jonathan,</FONT>
</P>

<P><FONT SIZE=3D2>Some initial responses are embedded below [MB].&nbsp; =
I do, however, have some more general comments that might also address =
your concerns as follows:</FONT></P>

<P><FONT SIZE=3D2>- In general, it seems that alot of your confusion =
around the fundamental processing of History-Info is around the fact =
that we've optimized to only capturing a single URI, which adds to some =
of the terminology concerns that you have. I'm beginning to wonder if =
it wouldn't be better to go back to capturing both the retargeted-from =
and retargeted-to URIs?&nbsp; The advantage of this is that it doesn't =
introduce this &quot;seeding&quot; or &quot;kickstarting&quot; of HI =
when a proxy receives a request with no HI, which introduces these =
special cases in the fundamental processing. What this also might be =
highlighting is the need to bring some of the background text on the =
solution options from appendix D into a background section that =
precedes the normative text for HI. </FONT></P>

<P><FONT SIZE=3D2>- As I mentioned during the discussion, I fully agree =
that there is a general lack of clarity (and as you suggest a need for =
more precision) around the fundmental normative processing after =
discussions with folks that are planning on implementing this draft. In =
particular, the processing around the Index is quite unclear; I think =
adding the index to all the examples and adding additional example =
flows will also help with the concern here. </FONT></P>

<P><FONT SIZE=3D2>Regards,</FONT>
<BR><FONT SIZE=3D2>Mary H. Barnes</FONT>
<BR><FONT SIZE=3D2>mbarnes@nortelnetworks.com</FONT>
</P>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Jonathan Rosenberg [<A =
HREF=3D"mailto:jdrosen@dynamicsoft.com">mailto:jdrosen@dynamicsoft.com</=
A>]</FONT>
<BR><FONT SIZE=3D2>Sent: Tuesday, July 15, 2003 7:58 PM</FONT>
<BR><FONT SIZE=3D2>To: sip@ietf.org</FONT>
<BR><FONT SIZE=3D2>Subject: [Sip] comments on history-info-00</FONT>
</P>
<BR>

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

<P><FONT SIZE=3D2>Here are some comments on history-info. I suspect =
many apply to prior </FONT>
<BR><FONT SIZE=3D2>versions as well. My apologies if they have been =
discussed, I have not </FONT>
<BR><FONT SIZE=3D2>really followed the work closely. Many of my =
comments are asking for </FONT>
<BR><FONT SIZE=3D2>much more precision in the document. Generally, I =
found it to be very </FONT>
<BR><FONT SIZE=3D2>imprecise, with lots of undefined terminology and =
vague procedures.</FONT>
</P>

<P><FONT SIZE=3D2>That aside, I have some major issues with the privacy =
mechanisms, with </FONT>
<BR><FONT SIZE=3D2>the usage of history-info by redirect servers, with =
sending a 183 with </FONT>
<BR><FONT SIZE=3D2>history information, and with the backwards =
compatibility </FONT>
<BR><FONT SIZE=3D2>implications. So, here we go:</FONT>
</P>

<P><FONT SIZE=3D2>Firstly, the draft says in many places that =
applications need to </FONT>
<BR><FONT SIZE=3D2>document the impact of things. For example, Section =
1.1 says:</FONT>
</P>

<P><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp; Thus, it is highly</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp; recommended that all applications =
making use of the request history</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp; information clearly define the =
impact of the information not being</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp; available and specify the =
processing of such a request.</FONT>
</P>

<P><FONT SIZE=3D2>Does this mean there is some kind of application =
registry where such </FONT>
<BR><FONT SIZE=3D2>documentation will exist? </FONT>
<BR><FONT SIZE=3D2>[MB]: No, I think the use of History-Info is no =
different than any other optional SIP header and that application =
scenarios in the form of call flows will be needed in Informational (or =
as has become the trend BCPs) to effectively document the use of the =
header. </FONT></P>

<P><FONT SIZE=3D2>Does it mean that applications won't be </FONT>
<BR><FONT SIZE=3D2>interoperable because they will depend on the =
presence of these </FONT>
<BR><FONT SIZE=3D2>headers? </FONT>
<BR><FONT SIZE=3D2>[MB]: Ditto my previous comment.</FONT>
</P>

<P><FONT SIZE=3D2>The text in 1.1 makes it sound like the information =
would </FONT>
<BR><FONT SIZE=3D2>only be present if some local policy prevented its =
insertion.</FONT>
<BR><FONT SIZE=3D2>[MB] I think you mean &quot;didn't prevent&quot; =
rather than &quot;prevented&quot; in this preceeding sentence.&nbsp; =
And, yes,local policy is one factor that determines whether the =
History-Info is inserted.</FONT></P>

<P><FONT SIZE=3D2>What about cases where the proxy doesn't support it? =
</FONT>
<BR><FONT SIZE=3D2>[MB]: Then, the information isn't captured for any =
retargeting by that proxy, but any previous HI entries should remain, =
since proxies should not remove headers. </FONT></P>

<P><FONT SIZE=3D2>Even if it does, your </FONT>
<BR><FONT SIZE=3D2>draft reads very much like it is now going to =
require every proxy to </FONT>
<BR><FONT SIZE=3D2>insert this header. Is that so?</FONT>
<BR><FONT SIZE=3D2>[MB]: No, it's an optional header. However, for =
systems where this isn't supported, the functionality of an end =
application might be limited. </FONT></P>

<P><FONT SIZE=3D2>Second, section 1.3 talks about the use of the =
Privacy header. There </FONT>
<BR><FONT SIZE=3D2>are a few problems here. First, the privacy header =
gets stripped once </FONT>
<BR><FONT SIZE=3D2>the privacy service finishes execution. Thus, a =
downstream proxy would </FONT>
<BR><FONT SIZE=3D2>not be able to know privacy was requested. To a =
large degree, this is </FONT>
<BR><FONT SIZE=3D2>because the privacy header in a request is about =
privacy of the </FONT>
<BR><FONT SIZE=3D2>caller. Request history is about the current target =
(i.e., called </FONT>
<BR><FONT SIZE=3D2>party). So, in fact, the issue is that if the =
*called* party requests </FONT>
<BR><FONT SIZE=3D2>privacy (using a privacy header in a response), the =
history info </FONT>
<BR><FONT SIZE=3D2>headers should not be propagated backwards in a =
response. I dont think </FONT>
<BR><FONT SIZE=3D2>your draft does that.</FONT>
</P>

<P><FONT SIZE=3D2>[MB]: This is similar to several threads of =
discussion we had around the requirements about 6 months ago:</FONT>
<BR><FONT SIZE=3D2><A =
HREF=3D"http://www1.ietf.org/mail-archive/working-groups/sipping/current=
/msg03695.html" =
TARGET=3D"_blank">http://www1.ietf.org/mail-archive/working-groups/sippi=
ng/current/msg03695.html</A></FONT>
<BR><FONT SIZE=3D2>which I believe was addressed in a subsequent =
revision to the requirements (although, I'll have to dig for the =
details on that as I don't see that readily visible in the archives or =
cached in my email archives at this =
time).&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT></P>

<P><FONT SIZE=3D2>Local policy might also play a role, when it is =
representing the </FONT>
<BR><FONT SIZE=3D2>privacy interests of a user that is retargeting. In =
that case, it was </FONT>
<BR><FONT SIZE=3D2>unclear to me how the proxy prevents downstream =
elements from </FONT>
<BR><FONT SIZE=3D2>inserting history-info? Your draft makes it sound =
like the privacy </FONT>
<BR><FONT SIZE=3D2>requirements are met if just this one hop avoids =
insertion. But seeing </FONT>
<BR><FONT SIZE=3D2>the contact from downstream retargets will often =
reveal sensitive </FONT>
<BR><FONT SIZE=3D2>information about a target further upstream.</FONT>
</P>

<P><FONT SIZE=3D2>[MB]: I agree, I think local policy is this primary =
mechanism for the privacy solution for HI and per the email =
discussion.&nbsp; As I highlighted in the meeting, the privacy aspects =
of the solution are indeed incomplete and hopefully will be addressed =
adequately in the next version. </FONT></P>

<P><FONT SIZE=3D2>Section 2.1:</FONT>
</P>

<P><FONT SIZE=3D2>&gt; Targeted-to-URI: the Request URI captured as the =
Request is </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
targeted. By capturing a copy of the Request URI in the initial </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
request, the Retargeted-from-URI is already captured when a </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
request is retargeted and the Retargeted-to-URI is being </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
captured.&nbsp;&nbsp; </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; </FONT>
</P>

<P><FONT SIZE=3D2>I read this many times and could not make sense of =
it. What does it mean?</FONT>
</P>

<P><FONT SIZE=3D2>[MB] This has to be read in context of the =
definitions for Retargeted-from-URI and Retargeted-to-URI in the =
Definitions section.&nbsp; Because we're optimizing by capturing only a =
single URI (the &quot;new&quot; URI in the request being retargeted) =
rather than capturing both the &quot;old&quot; and &quot;new&quot; URIs =
for each retargeting (and thus duplicating information since the =
&quot;new&quot; of the previous HI entry would become the =
&quot;old&quot; for the next entry). </FONT></P>

<P><FONT SIZE=3D2>Section 2.3.1:</FONT>
</P>

<P><FONT SIZE=3D2>&gt; The UAC SHOULD include the HistInfo option tag =
in the Supported </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; header in any request not =
associated with an established dialog for </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; which the UAC would like the =
History-Info in the Response.&nbsp; In </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; addition, the UAC should =
initiate the capturing of the History </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; Information by capturing the =
Request-URI as the hi-targeted-to-uri </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; and initializing the index to =
1.&nbsp; </FONT>
</P>

<P><FONT SIZE=3D2>What if the UAC doesn't support this extension? Can =
the UAS get </FONT>
<BR><FONT SIZE=3D2>history info? My understanding is that most, if not =
all, evnisioned </FONT>
<BR><FONT SIZE=3D2>applications were for the UAS. This means the UAS =
can't get </FONT>
<BR><FONT SIZE=3D2>information unless the caller and many, if not all, =
proxies along the </FONT>
<BR><FONT SIZE=3D2>path support it.</FONT>
</P>

<P><FONT SIZE=3D2>[MB] If the UAC doesn't include the option tag, then =
that means that it won't get the History-Info in any responses.&nbsp; =
Whether the proxy captures History-Info is local policy; the text is =
attempting to state that if the proxy is planning on capturing =
History-Info then it goes ahead and creates the &quot;zero-th&quot; =
entry.&nbsp; We originally were capturing 2 URIs, the original request =
URI and the retargeted-to URI. However, to minimize the overhead and =
duplicate data, we made the change in the -01 individual draft to only =
capture one URI, so that you basically need to &quot;seed&quot; the =
chain of URIs to capture the History, by capturing the original URI at =
the time of the first re-targeting that you're intending to =
capture.&nbsp; So, the UAS can get the information even if the caller =
doesn't want it in responses.&nbsp; You are correct that for many =
applications, it would be necessary for many of the proxies along the =
path to support it.&nbsp; </FONT></P>

<P><FONT SIZE=3D2>Section 2.3.2:</FONT>
</P>

<P><FONT SIZE=3D2>&gt; The processing of History-Info by a UAS in a =
Request depends upon </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; local policy and specific =
applications at the UAS which might make </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; use of the information.&nbsp; =
If the HistInfo option tag is received in a </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; request, the UAS should =
include any History-Info received in the </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; request in the subsequent resp=
onse.&nbsp;&nbsp;&nbsp;&nbsp; </FONT>
</P>

<P><FONT SIZE=3D2>I think you mean SHOULD in the last sentence. Also, I =
think you mean </FONT>
<BR><FONT SIZE=3D2>that &quot;If the request contained the histinfo =
option tag in the </FONT>
<BR><FONT SIZE=3D2>Supported header field of the request, the =
UAS....&quot;.</FONT>
</P>

<P><FONT SIZE=3D2>[MB] Correct.</FONT>
</P>

<P><FONT SIZE=3D2>Also, you've capitalized the H and I in HistInfo. =
Please don't. Option </FONT>
<BR><FONT SIZE=3D2>tags are not case sensitive.</FONT>
</P>

<P><FONT SIZE=3D2>[MB] Noted.</FONT>
</P>

<P><FONT SIZE=3D2>Section 2.3.3:</FONT>
</P>

<P><FONT SIZE=3D2>The wording about when to capture is really vague. =
You need to be much </FONT>
<BR><FONT SIZE=3D2>more precise in your wording of when the proxy is =
and isnt going to </FONT>
<BR><FONT SIZE=3D2>capture history information.</FONT>
</P>

<P><FONT SIZE=3D2>[MB]: As I mentioned during my presentation of issues =
during the WG meeting, this was the same feedback I had received from =
some folks that in the process of implementing this and I do improve =
this in the next version. However, this section was intended to be a =
higher level overview of the processing with the subsequent =
sub-sections intended to provide the details of the normative =
processing. </FONT></P>

<P><FONT SIZE=3D2>Section 2.3.3.1:</FONT>
</P>

<P><FONT SIZE=3D2>&gt;&nbsp; If the proxy supports History-Info, the =
proxy SHOULD add any History-</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; Info collected as it =
retargets a Request. For retargets that are the </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; result of an explicit SIP =
response, the SIP Response Code that </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; triggered the retargeting =
MUST be included in the Reason header of </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; the Targeted-to-URI.</FONT>
</P>

<P><FONT SIZE=3D2>This needs to be more precise. What does it mean =
&quot;..collected as it </FONT>
<BR><FONT SIZE=3D2>retargets a request&quot;? How is this information =
collected? Also, when </FONT>
<BR><FONT SIZE=3D2>you say &quot;explicit SIP response&quot; - all =
responses are explicit as far as </FONT>
<BR><FONT SIZE=3D2>I know. Perhaps you mean when a 3xx response is =
received? Also, you </FONT>
<BR><FONT SIZE=3D2>say &quot;Reason header of the =
Targeted-to-URI&quot;. I believe reason is a </FONT>
<BR><FONT SIZE=3D2>parameter, not a header, in this instance.</FONT>
</P>

<P><FONT SIZE=3D2>and then:</FONT>
<BR><FONT SIZE=3D2>&gt; For retargets as a result of timeouts or =
</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; internal events, a Reason =
header MAY be included in the Reason header </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; of the Targeted-to-URI. =
</FONT>
</P>

<P><FONT SIZE=3D2>what are internal events?</FONT>
</P>

<P><FONT SIZE=3D2>[MB]: This was also something I mentioned at the =
meeting and has to do with &quot;internal retargeting&quot; which can =
be the result of things like selecting the route in a route list, etc. =
******************************************</FONT></P>

<P><FONT SIZE=3D2>What value does the reason parameter have in the case =
of timeout? In </FONT>
<BR><FONT SIZE=3D2>the case of internal events?</FONT>
</P>

<P><FONT SIZE=3D2>[MB]: It lets an application or end user know why the =
retargeting occurred.</FONT>
</P>

<P><FONT SIZE=3D2>and then:</FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp; Additionally, if a request is received =
that doesn't include a </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; captured Request URI from the =
previous entity, the proxy MAY add an </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; additional entry, effectively =
capturing the retargeted-from-URI in </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; the Request.&nbsp;&nbsp; =
</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>What does it mean for a request to &quot;not include =
a captured </FONT>
<BR><FONT SIZE=3D2>Request-URI&quot;? You need to define that. What is =
contained in the </FONT>
<BR><FONT SIZE=3D2>additional entry?</FONT>
</P>

<P><FONT SIZE=3D2>[MB]: Per my general comment at the beginning of my =
email response, this has to do with the current mechanism of only =
capturing a single URI. </FONT></P>

<P><FONT SIZE=3D2>I may sound like I am being nit-picky, but in my =
experience </FONT>
<BR><FONT SIZE=3D2>specifications need to be very precise in what you =
want people to do. </FONT>
<BR><FONT SIZE=3D2>I am confused about what is implied here, and I =
would imagine many </FONT>
<BR><FONT SIZE=3D2>others would be confused too.</FONT>
</P>

<P><FONT SIZE=3D2>[MB]: Again, as I mentioned during the meeting, this =
is an open issue with the draft that I plan to correct on the next =
version.</FONT></P>

<P><FONT SIZE=3D2>and then:</FONT>
</P>

<P><FONT SIZE=3D2>&gt; In order to maintain ordering and accurately =
reflect the nesting and </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; retargeting of the request, =
an index MUST be included along with the </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; Targeted-to-URI being =
captured. The basic rule for adding the index </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; are to read the value from =
the previous History-Info, if available, </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; and capture the index.n as =
the index for the History-Info being </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; captured, where n would =
typically be 1 for a forwarded request. Thus, </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; the level of nesting of the =
index reflects the number of hops.</FONT>
</P>

<P><FONT SIZE=3D2>define previous.</FONT>
</P>

<P><FONT SIZE=3D2>what does it mean to &quot;capture the index.n&quot;? =
what is index? what is .n? </FONT>
<BR><FONT SIZE=3D2>&nbsp; You say its &quot;typically 1&quot;. When is =
it 1, and when is it not one?</FONT>
</P>

<P><FONT SIZE=3D2>and then:</FONT>
<BR><FONT SIZE=3D2>&gt; For </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; retargets within a proxy, the =
proxy MUST maintain the current level </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; of nesting by incrementing =
the lowest/last digit of the index for </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; each instance of retargeting, =
thus reflecting the number of retargets </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; within the proxy.</FONT>
</P>

<P><FONT SIZE=3D2>I cannot understand this at all.</FONT>
</P>

<P><FONT SIZE=3D2>and then:</FONT>
<BR><FONT SIZE=3D2>&gt; n index MUST NOT be added </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; in the scenario whereby the =
received request had no History-Info </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; header and the =
retargeted-from-URI is being captured for </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; completeness.</FONT>
</P>

<P><FONT SIZE=3D2>what does &quot;for completeness&quot; mean? You need =
to be precise.</FONT>
</P>

<P><FONT SIZE=3D2>[MB]: Another issue related to the general mechanism =
of capturing a single URI that I mentioned at the beginning of my email =
response. </FONT></P>

<P><FONT SIZE=3D2>&gt; The lack of Reason headers in the captured =
Request-URIs should be </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; indicative of the parallel =
nature of forking (i.e the Request-URIs </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; are not the result of =
retargets, but are rather all simultaneous </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; Targeted-To URIs.)&nbsp; =
</FONT>
</P>

<P><FONT SIZE=3D2>should be indicative? Is it, or isnt it?</FONT>
</P>

<P><FONT SIZE=3D2>[MB] Correct. I'll change this from &quot;should =
be&quot; to &quot;is&quot;</FONT>
</P>

<P><FONT SIZE=3D2>&gt; A proxy that receives a Request with the =
HistInfo option tag in the </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; Supported header, and =
depending upon a local policy supporting the </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; capture of History-Info, =
SHOULD return captured History-Info in </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; subsequent, provisional and =
final responses to the Request.&nbsp; A 183 </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; response MAY be sent =
explicitly for the purposes of conveying </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; History-Info prior to the =
final response. </FONT>
</P>

<P><FONT SIZE=3D2>I think the first sentence means that the proxy does =
nothing to </FONT>
<BR><FONT SIZE=3D2>responses - any history-info in those responses is =
just passed on. But </FONT>
<BR><FONT SIZE=3D2>I'm not sure. The sentence says &quot;should =
return&quot;, and return seems like </FONT>
<BR><FONT SIZE=3D2>an active operation to me. </FONT>
<BR><FONT SIZE=3D2>[MB]: The use of HI does not at all change the =
messaging, but rather just adds the captured information to the =
response in the scenarios where it has been captured. It is not =
intended to be an &quot;active&quot; operation.&nbsp; </FONT></P>

<P><FONT SIZE=3D2>The next sentence actually implies that the </FONT>
<BR><FONT SIZE=3D2>proxy generate a 183 provisional responses. </FONT>
<BR><FONT SIZE=3D2>[MB]: No, that's not what was intended.&nbsp; This =
sentence was added in response to a concern raised that HI didn't =
appear to be allowed to be sent in provisional responses.&nbsp; =
</FONT></P>

<P><FONT SIZE=3D2>This means that a UA </FONT>
<BR><FONT SIZE=3D2>making a call with N proxies will get N extra =
provisional responses, </FONT>
<BR><FONT SIZE=3D2>where the first has one history-info value, the next =
2, the next 3, </FONT>
<BR><FONT SIZE=3D2>and so on. The first history info is transmitted to =
the UAC N times. </FONT>
<BR><FONT SIZE=3D2>Is that really what we want? It seems very expensive =
for the UAC.</FONT>
<BR><FONT SIZE=3D2>[MB]: Per my response to a previous comment above, =
HI does not at all change the fundamental messaging, but rather just =
adds the captured information to the response in the scenarios where it =
has been captured.</FONT></P>
<BR>

<P><FONT SIZE=3D2>&gt; It MAY be advantageous for redirect servers to =
support the receipt of </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; History-Info in =
requests.</FONT>
</P>

<P><FONT SIZE=3D2>You can you assign a protocol strength to being =
advantageous?</FONT>
</P>

<P><FONT SIZE=3D2>I think you mean &quot;A redirect server MAY use =
information in the </FONT>
<BR><FONT SIZE=3D2>History-Info header field in a request to determine =
the URIs that it </FONT>
<BR><FONT SIZE=3D2>will retarget to.&quot;.</FONT>
</P>

<P><FONT SIZE=3D2>[MB] Agreed; your suggested wording is a more =
accurate representation of the intent of the original statement. =
</FONT>
</P>

<P><FONT SIZE=3D2>&gt; By receiving it in the request, the </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; Redirect Server MAY be able =
to optimize the information it sends in </FONT>
<BR><FONT SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp; responses by looking at the =
already targeted-to-URIs.</FONT>
</P>

<P><FONT SIZE=3D2>and what optimization is that? I think you mean that =
the redirect </FONT>
<BR><FONT SIZE=3D2>server shouldnt redirect to somewhere that the =
request has already </FONT>
<BR><FONT SIZE=3D2>visited. If so, please say that. However, are we =
sure about that? </FONT>
<BR><FONT SIZE=3D2>Redirection to a URI already visited can be =
appropriate in cases where </FONT>
<BR><FONT SIZE=3D2>there is a spiral, and not a loop. Unless a redirect =
server is </FONT>
<BR><FONT SIZE=3D2>maintaining state, how can it make this =
determiantionabout whether </FONT>
<BR><FONT SIZE=3D2>this is a spiral or a loop?</FONT>
</P>

<P><FONT SIZE=3D2>[MB]: You bring up a valid point that should be =
clarified.&nbsp; It might be better to just remove that statement since =
it really is an implementation decision and as you suggest such =
&quot;optimizations&quot; could be problematic in some scenarios.&nbsp; =
(Although, it might be better to highlight the cautions should someone =
want to make such an optimization or highlight perhaps that it's better =
not to do such an optimization). </FONT></P>
<BR>
<BR>

<P><FONT SIZE=3D2>&gt; HistInfo&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; When used =
with the Supported header, [RFCXXXX] </FONT>
<BR><FONT =
SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; this option tag indicates =
support </FONT>
<BR><FONT =
SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; for the History Information =
to be&nbsp; </FONT>
<BR><FONT =
SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; captured for requests and =
returned in </FONT>
<BR><FONT =
SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; subsequent responses. This =
tag is not </FONT>
<BR><FONT =
SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; used in a Proxy-Require or =
Requires&nbsp; </FONT>
<BR><FONT =
SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; header field since support =
of&nbsp; </FONT>
<BR><FONT =
SIZE=3D2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; History-Info is =
optional.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Require header field, not Requires.</FONT>
</P>

<P><FONT SIZE=3D2>[MB]: Noted. </FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Thanks,</FONT>
<BR><FONT SIZE=3D2>Jonathan R.</FONT>
</P>

<P><FONT SIZE=3D2>-- </FONT>
<BR><FONT SIZE=3D2>Jonathan D. Rosenberg, =
Ph.D.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp; 600 Lanidex Plaza</FONT>
<BR><FONT SIZE=3D2>Chief Technology =
Officer&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Parsippany, NJ =
07054-2711</FONT>
<BR><FONT SIZE=3D2>dynamicsoft</FONT>
<BR><FONT =
SIZE=3D2>jdrosen@dynamicsoft.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp; FAX:&nbsp;&nbsp; (973) 952-5050</FONT>
<BR><FONT SIZE=3D2><A HREF=3D"http://www.jdrosen.net" =
TARGET=3D"_blank">http://www.jdrosen.net</A>&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp; PHONE: (973) 952-5000</FONT>
<BR><FONT SIZE=3D2><A HREF=3D"http://www.dynamicsoft.com" =
TARGET=3D"_blank">http://www.dynamicsoft.com</A></FONT>
</P>
<BR>
<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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Hi,
Can anybody tell me abt sip implementation framework.

Manish

--
Things work out best for those who make the best use of it.



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Hi,
<br>Can anybody tell me abt sip implementation framework.
<p>Manish
<pre>--&nbsp;
Things work out best for those who make the best use of it.</pre>
&nbsp;</html>

--------------6694E02067B363CACD1A716E--



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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		: Communications Resource Priority for the Session 
                          Initiation Protocol (SIP)
	Author(s)	: H. Schulzrinne, J. Polk
	Filename	: draft-ietf-sip-resource-priority-01.txt
	Pages		: 24
	Date		: 2003-7-28
	
This document defines two new SIP header fields for communications
resource priority, namely 'Resource-Priority' and 'Accept-Resource-
Priority'. The Resource-Priority header field can influence the
behavior of SIP UAs, such as GSTN gateways, and SIP proxies. It does
not directly influence the forwarding behavior of IP routers.

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From: Henning Schulzrinne <hgs@cs.columbia.edu>
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The revised version of the draft, based on discussions in Vienna, is now 
in the archives. I would appreciate comments

http://www.ietf.org/internet-drafts/draft-ietf-sip-resource-priority-01.txt

Thanks.

Henning


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I THINK I have all slides from the recent SIP WG meeting posted at:

http://www.softarmor.com/sipwg/meets/ietf57/slides/index.html


If you presented slides please check and make sure they're there.

If I'm missing something please let me know.


Thanks,

--
Dean





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dan,

thanks for the comments.  see responses/questions inline...


Romascanu, Dan (Dan) wrote:
> Here are my comments concerning the latest version of the SIP MIB. I 
> congratulate the authors for a much better document, and I thank them 
> for taking into account and including fixes to most of the problems that 
> were included in the review I did on the previous version.
> 
> There are still a few problems that need to be fixed:
> 
> A1) smilint complains about the following:
> 
> SIP-COMMON.txt:2014: [5] {integer-misuse} warning: use Integer32 instead 
> of INTEGER in SMIv2 -- for sipStatusCodeValue --

done.  used Unsigned32 actually as the range is postive.

> 
> SIP-COMMON.txt:2940: [5] {group-unref} warning: current group 
> `sipCommonNotifObjectsGroup' is not referenced in this module

i presume adding a GROUP clause to the module-compliance will
fix this.   i can do that.   however, my understanding is that
if a group is not mentioned, it's considered optional.
at any rate, i'll add the GROUP clause - as there's one for
each other optional group.


> 
> A2) You would like to re-consider the OID place holders that you created 
> for sipRedirCfg, sipRedirStats, sipRegCfg, sipRegStats.
> 
> This is not an SMI violation, but it creates for the time being a hole 
> in a MIB walk. If you need to add the definition of the respective 
> objects in the future, you will not be able to do it in the same 
> document, without the need to re-cycle the document at Proposed Standard 
> level.

ok.  as this has come up at least a couple of times now, i'll remove them.


> 
> A3) DESCRIPTION Clause of sipServiceNotifEnable -  Saying that in case 
> notifications are not implemented, the value has 'no meaning and MUST be 
> 0', seems contradictory and inaccurate. The respective sentence should 
> read - 'The notifications are OPTIONAL, and if they are not implemented, 
> this object's value MUST be all 0s.'
> 
> A4) How does sipNotifSequenceNumber work? It should probably be 
> increased once for each notification defined in the SIP MIBs, but not 
> for each notification target (in the case of multiple managers). I 
> suggest this to be detailed in the DESCRIPTION clause for clarity.
> 
>  
> 
> Minor editorial issues:
> 
> B1) English syntax in section 4.1. It should probably read 'The data 
> type SipTransportProtocol is used as Textual Convention in this 
> document. Textual conventions have...'

done.


> 
> B2) Eliminate the commented text in the SIP compliance module of the UA 
> MIB. 

you mean where it says "SMIC compiler problems"?

realize that OBJECT refinement was added in response to one of your
previous comments on rev -05...

    "B9 - If only a subset of the values of a TC is supported, like in
     the case of sipUACfgSipServerAddrStatus, this needs to be reflected
     in the compliance clauses. See Section 4.8 of 
http://www.ietf.org/internet-drafts/draft-ietf-ops-mib-review-guidelines-01.txt
for an example"

so remove it completely??

kevin

> 
> Regards,
> 
> Dan
> 


-- 
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
  Kevin R. Lingle       919.392.2029
  http://www.klove.com                               http://www.air1.com
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one mistake...

Romascanu, Dan (Dan) wrote:

<snip>
> A2) You would like to re-consider the OID place holders that you created 
> for sipRedirCfg, sipRedirStats, sipRegCfg, sipRegStats.

sipRegCfg and sipRegStats aren't placeholders.  they have actual
objects defined under them.

i'll remove the redirect related placeholders though.

kevin
-- 
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
  Kevin R. Lingle       919.392.2029
  http://www.klove.com                               http://www.air1.com
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dan, other question...

Kevin Lingle wrote:

>>
>> B2) Eliminate the commented text in the SIP compliance module of the 
>> UA MIB. 
> 
> 
> you mean where it says "SMIC compiler problems"?

if you do mean this... there is an occurrence in the SIP-COMMON-MIB
as well.   remove it as well??

kevin

> 
> realize that OBJECT refinement was added in response to one of your
> previous comments on rev -05...
> 
>    "B9 - If only a subset of the values of a TC is supported, like in
>     the case of sipUACfgSipServerAddrStatus, this needs to be reflected
>     in the compliance clauses. See Section 4.8 of 
> http://www.ietf.org/internet-drafts/draft-ietf-ops-mib-review-guidelines-01.txt 
> 
> for an example"
> 
> so remove it completely??
> 
> kevin
> 
>>
>> Regards,
>>
>> Dan
>>
> 
> 


-- 
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
  Kevin R. Lingle       919.392.2029
  http://www.klove.com                               http://www.air1.com
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Subject: [Sip] RFC 3319 on Dynamic Host Configuration Protocol (DHCPv6) Options for Session Initiation Protocol (SIP) Servers
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--NextPart


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


        RFC 3319

        Title:      Dynamic Host Configuration Protocol (DHCPv6)
                    Options for Session Initiation Protocol (SIP)
                    Servers
        Author(s):  H. Schulzrinne, B. Volz
        Status:     Standards Track
        Date:       July 2003
        Mailbox:    schulzrinne@cs.columbia.edu,
                    volz@metrocast.net
        Pages:      7
        Characters: 14444
        Updates/Obsoletes/SeeAlso:  None

        I-D Tag:    draft-ietf-sip-dhcpv6-01.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3319.txt


This document defines a Dynamic Host Configuration Protocol version 6
(DHCPv6) option that contains a list of domain names or IPv6 addresses
that can be mapped to one or more Session Initiation Protocol (SIP)
outbound proxy servers.  This is one of the many methods that a SIP
client can use to obtain the addresses of such a local SIP server.

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
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        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
Content-Type:  Message/External-body;
        access-type="mail-server";
        server="RFC-INFO@RFC-EDITOR.ORG"

Content-Type: text/plain
Content-ID: <030730163622.RFC@RFC-EDITOR.ORG>

RETRIEVE: rfc
DOC-ID: rfc3319

--OtherAccess
Content-Type:   Message/External-body;
        name="rfc3319.txt";
        site="ftp.isi.edu";
        access-type="anon-ftp";
        directory="in-notes"

Content-Type: text/plain
Content-ID: <030730163622.RFC@RFC-EDITOR.ORG>

--OtherAccess--
--NextPart--

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