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Subject: In-band soft-state control plane?RE: [Ieprep] I-D ACTION:draft-ietf-ieprep-domain-req-00.txt
To: "James M. Polk" <jmpolk@cisco.com>
Cc: ieprep@ietf.org, ieprep-admin@ietf.org
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From: Janet P Gunn <jgunn6@csc.com>
Date: Mon, 2 Feb 2004 12:02:44 -0500
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James,

Can you explain to me exactly what is meant by  "In_band soft_state control
plane ".  It sounds great but I am having difficulty envisioning exactly
what it is. (Or refer me to the appropriate section of an ID or RFC)

Thanks



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                      "James M. Polk"                                                                                            
                      <jmpolk                  To:      "Ken Carlberg" <carlberg@g11.org.uk>                                     
                      @cisco.com>              cc:      <ieprep@ietf.org>                                                        
                      Sent by:                 Subject: RE: [Ieprep] I-D ACTION:draft-ietf-ieprep-domain-req-00.txt              
                      ieprep-admin                                                                                               
                                                                                                                                 
                                                                                                                                 
                      01/29/2004 06:04                                                                                           
                      PM                                                                                                         
                                                                                                                                 
                                                                                                                                 



...
It is Fred Baker's and my opinion that an In_band soft_state control plane
would be the very best of all worlds to provide all manners of offered load

to capacity analysis at the congestion point - and not continue to rely on
some server architecture to maintain the congestive state of all interfaces

for 1000s of router interfaces (in rear-time).

...

cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


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Subject: RE: [Ieprep] I-D ACTION:draft-ietf-ieprep-domain-req-00.txt
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Picky is fine.  I have been known to be picky!

Remaining comments in line.


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                      "Ken Carlberg"                                                                                             
                      <carlberg                To:      "'Janet P Gunn'" <jgunn6@csc.com>                                        
                      @g11.org.uk>             cc:      <ieprep@ietf.org>                                                        
                                               Subject: RE: [Ieprep] I-D ACTION:draft-ietf-ieprep-domain-req-00.txt              
                      01/28/2004 01:40                                                                                           
                      PM                                                                                                         
                                                                                                                                 
                                                                                                                                 




Janet,

The following response may be a bit picky on some issues/comments, but I
feel it will be helpful to clarify things a bit.

> I think that, under "normal" circumstances, the QoS mechanisms for
> conventional VoIP will be more than adequate to support ETS, and
"special
> treatment" in Call Admission Control is what will be needed for ETS.

Within the context of what we have written in the past in the IETF, ETS
is an umbrella term that is meant to cover a wide variety of
applications.  Some of these applications (eg, I-Am-Alive) do not
support the concept of CAC. So to say that "special treatment" (whatever
that is) of CAC is what will be needed for ETS is rather too specific a
statement to make.

I think we can agree that in those applications used to support ETS and
dependent on CAC, an ability to distinguish and perform CAC is of high
interest and needs to be supported.

[jpg] you are right,  I should have said voice (and video) ETS
applications. Which prompts me to ask if there will be a single domain IP
telephony ETS requirements document.

> As Fred Baker says in draft-baker-tsvwg-mlpp-that-works.00.txt, if
call
> admission is working properly, traffic marking is not needed in normal
> operations.  It is when there are route changes - typically due to
network
> failures - that marking is needed.

Keep in mind that Fred's draft is in respect to MLPP.  Arguments from
the draft may be applied to other circumstances, but I would refrain
from treating it as a blanket statement for all applications and
situations.

[jpg] Agree, not a blanket statement. But my point is that when something
happens to/in the network, that is the time when "extraordinary" treatment
may make sense (recognizing that even without extraordinary treatment, this
is still vastly superior to the PSTN situation, where ANY disruption to the
network (on the particular call path) makes the call go away).  This is
particularly true for voice and video applications.  Less so, I agree, for
elastic applications.

 I've worked on systems in the past where "marking" was used
to separate traffic into either different physical subnets or different
virtuals paths -- neither case where the route changes existed because
of network failures.

> When there are network failures, the
> capacity and route assignments on which call admission was based no
longer
> apply, the QoS cannot be guaranteed until the new routes are
stabilized,
> and the call admission process adapts to the new capacity and routes.
> This may be a very local effect, of quite short duration.  But it has
a
> significant impact on the call quality of the calls that are affected.
> While that paper addresses an IP implementation of MLPP, many of the
same
> concerns apply to ETS.

Ok

> Network failures, and/or  massive and sustained statistical anomalies
> (large, local bursts), and the associated route changes, are the
problems
> we need to focus on.
>
> It is in this context that ETS calls must be given a high probability
of
> meeting the QoS criteria, even when network congestion and or damage
are
> severe.

Ok.

> The requirements in draft-ietf-ieprep-domain-req-00.txt don't seem to
> address this, and the framework in
draft-ietf-ieprep-domain-frame-00.txt
> doesn't seem to solve it.

Actually, that's fine.  The former draft is not aimed at a specific
application or system (this is stated in the draft).  It is admittedly
broad and acts as a foundation for the context of a single domain.  The
latter draft is not meant to offer solutions, but rather to discuss
applicable and related protocols that one *could* encounter within the
specific scope of a single domain.  I will expand this below where you
have a specific question.

[jpg] I guess my unwritten question here is "Are you planning to have a
document which DOES address this."  I agree that it is somewhat
out-of-scpoe for the two new documents.

> In draft-gunn-ieprep-ip-telephony-gap-00.txt we address the  need to
> support ETS SLAs to provide High Probability Toll-Quality Voice - ETS
> calls
> must be given adequate resources through the network to ensure a high
> probability that their packets suffer loss, delay, and jitter
consistent
> with achieving toll-quality voice even when network congestion and /
or
> damage are severe.

<snip>

> What is the appropriate way to reflect these concerns?

Through a draft (and I state it with no intention of being obnoxiously
obvious). If your question is, how does one deal with the three
seemingly related drafts?
   1. draft-ietf-ieprep-domain-req-00.txt,
   2. draft-ietf-ieprep-domain-frame-00.txt, and
   3. draft-gunn-ieprep-ip-telephony-gap-00.txt,

then I would follow the position I stated above and have (3) simply be a
more specific continuation of what has been stated in (1) and (2) above.
In my view it may be more constructive to be very specific and identify
specific systems/efforts like GETS or NS/EP or WPS.  If the chairs or AD
feel differently, I'm sure they'll chime in.

[jpg] Thanks, that is encouraging.

Janet

Cheers,

-ken






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Subject: References to MLPP that works Re: [Ieprep] I-D ACTION:draft-ietf-ieprep-domain-req-00.txt
To: Fred Baker <fred@cisco.com>
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                      Fred Baker <fred                                                                                           
                      @cisco.com>              To:      Janet P Gunn <jgunn6@csc.com>                                            
                                               cc:      ieprep@ietf.org                                                          
                      01/27/2004 09:35         Subject: Re: [Ieprep] I-D ACTION:draft-ietf-ieprep-domain-req-00.txt              
                      PM                                                                                                         
                                                                                                                                 
                                                                                                                                 




At 12:32 PM 1/27/2004, Janet P Gunn wrote:
>As Fred Baker says in draft-baker-tsvwg-mlpp-that-works.00.txt, if call
>admission is working properly, traffic marking is not needed in normal
>operations.

lest I be misunderstood, let me restate that. What I wrote was:

    One will ask the value of the multiple DSCPs. They are, in fact, of
    limited to no value in normal operation, as all vice and video
    traffic should have been admitted, and therefore capacity will have
    been assigned to them. The behavior of this service will be
    indistinguishable from the EF PHB regardless of traffic marking.

In retrospect, I wonder what the vice traffic would be :^)

[jpg] Maybe all that "organ enlargement" spam?

This is not to be understood as saying "no traffic marking is required";
one still needs RFC 3246. But if bandwidth admission is operating
correctly, having N different marks will be operationally indistinguishable
from having a single EF mark, as the total number of bytes on the wire will
be the same.
jpg], I did understand that.  I was oversimplifying to make my point.

>It is in this context that ETS calls must be given a high probability of
>meeting the QoS criteria, even when network congestion and or damage are
>severe.

I wonder if we are communicating here. RFC 3246 is designed to operate in a
congested network, and to operate in a network in which routing may change.
The comment in the mlpp-that-works draft - which if it is giving cause for
confusion I will happily remove - is basically me bending over backwards to
be intellectually honest and see if I can find a case where CAC may be
insufficient.

[jpg] And I was blatantly taking advantage of your "bending over backwards"
- to identify a case where CAC may not be sufficient for the ETS Voice
application.

Think about this for a moment: the argument being placed for MLEF is that
even in the presence of some loss, voice is intelligible. If that is true,
then just after a route change, during the brief interval between the data
flow being switched to a new path and the CAC following that and either
OKing the flow or deciding to shut down the call, when there may be some
loss, voice should be equally intelligible, no?

 Or does the argument that voice is able to work around a modest level of
loss only work when it comes from the proponents of multiple code points?

[jpg] In the ETS voice scenario, I am NOT trying to say that "voice is able
to work around a modest level of loss".  I am taking the highly elitist
position that, if there is a situation where SOME "calls" are going to have
to lose packets, I want it to be the non-ETS calls (whether or not the
non-ETS calls remain intelligible or not) that lose packets, and not the
ETS calls.

[jpg] I further agree that, in any situation where "calls" are losing
packets, it is highly desirable to "do something" so that you have some
calls that remain up and intelligible, and so that other calls go away,
rather than using up bandwidth sending packets that are not "useful" from
the end user's perspective.  And whatever that "something" is, I want it to
be able (assuming the appropriate business agreements are in place) to
slant the odds in favor of the ETS calls being the ones that stay up and
intelligible.


Janet
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At 09:02 AM 2/2/2004, Janet P Gunn wrote:
>Can you explain to me exactly what is meant by  "In_band soft_state 
>control plane".

RSVP is an example of one. 
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Comments in line- trying to make sure I understand what you are saying.


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                      "James M. Polk"                                                                                            
                      <jmpolk                  To:      "Steve Silverman" <steves@shentel.net>, "Fred Baker" <fred@cisco.com>,   
                      @cisco.com>              Janet P Gunn/FED/CSC@CSC, <ieprep@ietf.org>                                       
                                               cc:                                                                               
                      01/29/2004 01:25         Subject: RE: [Ieprep] I-D ACTION:draft-ietf-ieprep-domain-req-00.txt              
                      PM                                                                                                         
                                                                                                                                 
                                                                                                                                 




comments below

At 11:33 AM 1/29/2004 -0500, Steve Silverman wrote:
>If the offered load exceeds the network capacity, some packets will be
>dropped.  If a line is at
>95% of capacity, a 10% surge would cause 5% of the packets to be
>dropped, degrading most of the conversations.  ( I am simplifying here
>but I think this is clear.)
>
>The benefit of multiple DSCPs is that the sessions designated most
>critical are unaffected.

without active feedback to those lower priority sessions....

>Actual experiment has shown that this does
>work.

yes it does - but without regard to any call quality.

If the surge is 25 or 50 or 75% above maximum load possible, then what
happens to the lower priority calls? There isn't any feedback to the users
(other than pure silence). Further, there isn't any feedback (signaling) to

their devices what's happening or to hang up the call. There's nothing. How

long is this expected to last for? No one knows.

Further, no one can test the system (because of no feedback).

No one can point to where there might be potential problems with this
architecture either (is the system behaving correctly?). There is no
feedback to discover or monitor its performance. Where is it supposed to be

discovered that a link between any two routers isn't fast enough (because
there are perhaps frequent silence periods)?


>In draft-baker-tsvwg-mlpp-that-works.00 it is assumed that the CAC
>will ensure that voice is not overloaded.
>Two problems with this are:
>
>         1)  If silence suppression is used and the bandwidth
"oversubscribed"
>(standard practice) then sometimes, more people on one end will talk
>and cause congestion.
>
>         2)  If the area controlled by the CAC is large, propagation delay
>will limit the reaction time of the system

where's the propagation delay? the surge won't happen without the signaling

having already occurred through the congestion point; meaning the system
(read: routers) already know about all the call (flow) attempts - even when

there is too much offered load for an interface. The router can either say
"sorry, not enough resources available to complete your call attempt" or
"sorry, but there was another call with higher priority, and I have to shut

your call down now". BTW - these are device signals, this isn't a .wav file

playing to the user or anything.

[jpg] I think I understand how CAC says "not enough resources available to
complete your call attempt" at call setup time.

[jpg] It is not clear to me how CAC says "sorry, but there was another call
with higher priority, and I have to shut your call down now" once the call
has been established.  I was under the impression that, once the call was
established, the router did not distinguish between individual calls.  I
would be glad to be enlightened on that point.

>and the computation to
>oversee the entire theatre may also be a limiting factor.

see above, computation occurs at the router that is experiencing this
surge. It is in charge of the packet forwarding. It is aware of all the
existing flows it has accepted. It is in charge of accepting new flows and
preempting lesser priority ones if multilevel offered load exceeds the
maximum load possible.

>This may
>cause short congestion incidents (several seconds long) which could be
>mitigated by MLEF.  I'm not saying you don't also need the CAC.  I'm
>saying
>you need something local and fast reacting

In the router itself isn't fast enough...?

>(at the packet level rather
>than the call level)

packet level = Layer 3

Call level = what? Layer 4, 5 or 7?

>to ensure that the high priority packets get
>thru.

I am a believer in high priority calls getting through, and for all calls
that remain up - having all their packets arrive safely too.

[jpg] Me too

Janet


>Steve Silverman


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented






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Subject: Re: In-band soft-state control plane?RE: [Ieprep] I-D  ACTION:draft-ietf-ieprep-domain-req-00.txt
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Thanks- clearer now.


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                      Fred Baker <fred                                                                                           
                      @cisco.com>              To:      Janet P Gunn/FED/CSC@CSC                                                 
                                               cc:      ieprep@ietf.org                                                          
                      02/02/2004 05:02         Subject: Re: In-band soft-state control plane?RE: [Ieprep] I-D                    
                      PM                       ACTION:draft-ietf-ieprep-domain-req-00.txt                                        
                                                                                                                                 
                                                                                                                                 




At 09:02 AM 2/2/2004, Janet P Gunn wrote:
>Can you explain to me exactly what is meant by  "In_band soft_state
>control plane".

RSVP is an example of one.

<< Attachment Removed : C.DTF >>




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Date: Mon, 02 Feb 2004 16:57:51 -0600
To: Janet P Gunn <jgunn6@csc.com>
From: "James M. Polk" <jmpolk@cisco.com>
Subject: Re: CAC RE: [Ieprep] I-D
  ACTION:draft-ietf-ieprep-domain-req-00.txt
Cc: "Fred Baker" <fred@cisco.com>, ieprep@ietf.org,
        "Steve Silverman" <steves@shentel.net>
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Janet

My comparison in the quotes was a poor attempt at what occurs within MLPP 
Switches today (preemption tone or an announcement), and from an IP POV - 
the tone behavior is doable with my recent ID into SIPPING (Reason Header 
for Preemption), but the latter (the announcement) isn't doable to inform a 
user with words about which router was the culprit in not letting a call 
get through or the preemptor of a call.

The GSCR clearly call out these two behaviors, and they are probably 
appropriate for TDM networks, but IMO are not for IP networks where the 
congestion event happened at a router - but this distinction isn't made in 
the GSCR, both behaviors are still mandated (and I don't know how we're 
going to accomplish the announcement regarding a specific identifiable 
router causing a call to fail on purpose).

If this is an "IP at the Start" scenario, and the congestion event occurs 
within the TDM infrastructure, then I believe that announcement or tone 
MUST be communicated back to the IP phone and user on that call to inform 
them of what they get today

I know this is (slightly) off topic for this list


At 05:25 PM 2/2/2004 -0500, Janet P Gunn wrote:

>Comments in line- trying to make sure I understand what you are saying.
>
>
>----------------------------------------------------------------------------------------
>
>This is a PRIVATE message. If you are not the intended recipient, please
>delete without copying and kindly advise us by e-mail of the mistake in
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>bind CSC to any order or other contract unless pursuant to explicit written
>agreement or government initiative expressly permitting the use of e-mail
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>
>
>
> 
>
>                       "James M. 
> Polk" 
>
>                       <jmpolk                  To:      "Steve Silverman" 
> <steves@shentel.net>, "Fred Baker" <fred@cisco.com>,
>                       @cisco.com>              Janet P Gunn/FED/CSC@CSC, 
> <ieprep@ietf.org>
>                                                cc: 
 >
>                       01/29/2004 01:25         Subject: RE: [Ieprep] I-D 
> ACTION:draft-ietf-ieprep-domain-req-00.txt
>                       PM 
 >
> 
>
> 
>
>
>
>
>
>comments below
>
>At 11:33 AM 1/29/2004 -0500, Steve Silverman wrote:
> >If the offered load exceeds the network capacity, some packets will be
> >dropped.  If a line is at
> >95% of capacity, a 10% surge would cause 5% of the packets to be
> >dropped, degrading most of the conversations.  ( I am simplifying here
> >but I think this is clear.)
> >
> >The benefit of multiple DSCPs is that the sessions designated most
> >critical are unaffected.
>
>without active feedback to those lower priority sessions....
>
> >Actual experiment has shown that this does
> >work.
>
>yes it does - but without regard to any call quality.
>
>If the surge is 25 or 50 or 75% above maximum load possible, then what
>happens to the lower priority calls? There isn't any feedback to the users
>(other than pure silence). Further, there isn't any feedback (signaling) to
>
>their devices what's happening or to hang up the call. There's nothing. How
>
>long is this expected to last for? No one knows.
>
>Further, no one can test the system (because of no feedback).
>
>No one can point to where there might be potential problems with this
>architecture either (is the system behaving correctly?). There is no
>feedback to discover or monitor its performance. Where is it supposed to be
>
>discovered that a link between any two routers isn't fast enough (because
>there are perhaps frequent silence periods)?
>
>
> >In draft-baker-tsvwg-mlpp-that-works.00 it is assumed that the CAC
> >will ensure that voice is not overloaded.
> >Two problems with this are:
> >
> >         1)  If silence suppression is used and the bandwidth
>"oversubscribed"
> >(standard practice) then sometimes, more people on one end will talk
> >and cause congestion.
> >
> >         2)  If the area controlled by the CAC is large, propagation delay
> >will limit the reaction time of the system
>
>where's the propagation delay? the surge won't happen without the signaling
>
>having already occurred through the congestion point; meaning the system
>(read: routers) already know about all the call (flow) attempts - even when
>
>there is too much offered load for an interface. The router can either say
>"sorry, not enough resources available to complete your call attempt" or
>"sorry, but there was another call with higher priority, and I have to shut
>
>your call down now". BTW - these are device signals, this isn't a .wav file
>
>playing to the user or anything.
>
>[jpg] I think I understand how CAC says "not enough resources available to
>complete your call attempt" at call setup time.
>
>[jpg] It is not clear to me how CAC says "sorry, but there was another call
>with higher priority, and I have to shut your call down now" once the call
>has been established.  I was under the impression that, once the call was
>established, the router did not distinguish between individual calls.  I
>would be glad to be enlightened on that point.
>
> >and the computation to
> >oversee the entire theatre may also be a limiting factor.
>
>see above, computation occurs at the router that is experiencing this
>surge. It is in charge of the packet forwarding. It is aware of all the
>existing flows it has accepted. It is in charge of accepting new flows and
>preempting lesser priority ones if multilevel offered load exceeds the
>maximum load possible.
>
> >This may
> >cause short congestion incidents (several seconds long) which could be
> >mitigated by MLEF.  I'm not saying you don't also need the CAC.  I'm
> >saying
> >you need something local and fast reacting
>
>In the router itself isn't fast enough...?
>
> >(at the packet level rather
> >than the call level)
>
>packet level = Layer 3
>
>Call level = what? Layer 4, 5 or 7?
>
> >to ensure that the high priority packets get
> >thru.
>
>I am a believer in high priority calls getting through, and for all calls
>that remain up - having all their packets arrive safely too.
>
>[jpg] Me too
>
>Janet
>
>
> >Steve Silverman
>
>
>cheers,
>James
>
>                                 *******************
>                  Truth is not to be argued... it is to be presented


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


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> [jpg] I guess my unwritten question here is "Are you planning to have
a
> document which DOES address this."  I agree that it is somewhat
> out-of-scpoe for the two new documents.

My plans with respect to potential IEPREP documents is to work within
the scope of the charter.  Lets discuss this offline to make sure I know
exactly what you are looking for and I'll see if I can provide
assistance.

-ken



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> [jpg] It is not clear to me how CAC says "sorry, but there was another call
> with higher priority, and I have to shut your call down now" once the call
> has been established.  I was under the impression that, once the call was
> established, the router did not distinguish between individual calls.  I
> would be glad to be enlightened on that point.

note that the router is not involved in establishing the call - its all
packets to the router

but if the call QoS is maintained by RSVP (as Fred suggested) there is 
less of an issue - a router can reject RSVP sessions even after they
have started

Scott

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Subject: RE: References to MLPP that works Re: [Ieprep] I-D ACTION:draft-ietf-ieprep-domain-req-00.txt
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Janet and all

I have been biting my tongue (ops fingers) for some time now with
regards to the MLPP service for supporting ETS in IP networks. As this
service has a potential problem as shown by the following scenario:

If a call/session is in progress between a non authorised ETS user and a
emergency service then this should never be subject to pre-emption by a
authorised ETS user, it could have fatal consequences. This by extension
could be applied to a call/session in progress between non authorised
ETS users, as this contact could be the only point of hope for someone
in trouble.

In other words the only justification for a pre-emption by a authorised
ETS user is that it is only applied to call/session if both the content
and context have been analysed. This seems to be a lot of work in a
congested network.

I have no problems with multiple precedence's or priorities if they are
required but I only see the need for two a 'normal' call/session and a
'ETS' call/session.=20

This shows that there maybe a need to take into account the consequences
of the requirement to support MLPP especially in regard to pre-emption.=20

Regards

Steve Norreys =20
Signalling and Protocols
Tel:       +441277323220
email    steve.norreys@bt.com


-----Original Message-----
From: ieprep-admin@ietf.org [mailto:ieprep-admin@ietf.org] On Behalf Of
Janet P Gunn
Sent: 02 February 2004 22:13
To: Fred Baker
Cc: ieprep@ietf.org
Subject: References to MLPP that works Re: [Ieprep] I-D
ACTION:draft-ietf-ieprep-domain-req-00.txt



Comments in line.


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=20

                      Fred Baker <fred

                      @cisco.com>              To:      Janet P Gunn
<jgunn6@csc.com>                                           =20
                                               cc:      ieprep@ietf.org

                      01/27/2004 09:35         Subject: Re: [Ieprep] I-D
ACTION:draft-ietf-ieprep-domain-req-00.txt             =20
                      PM

=20

=20





At 12:32 PM 1/27/2004, Janet P Gunn wrote:
>As Fred Baker says in draft-baker-tsvwg-mlpp-that-works.00.txt, if call

>admission is working properly, traffic marking is not needed in normal=20
>operations.

lest I be misunderstood, let me restate that. What I wrote was:

    One will ask the value of the multiple DSCPs. They are, in fact, of
    limited to no value in normal operation, as all vice and video
    traffic should have been admitted, and therefore capacity will have
    been assigned to them. The behavior of this service will be
    indistinguishable from the EF PHB regardless of traffic marking.

In retrospect, I wonder what the vice traffic would be :^)

[jpg] Maybe all that "organ enlargement" spam?

This is not to be understood as saying "no traffic marking is required";
one still needs RFC 3246. But if bandwidth admission is operating
correctly, having N different marks will be operationally
indistinguishable from having a single EF mark, as the total number of
bytes on the wire will be the same. jpg], I did understand that.  I was
oversimplifying to make my point.

>It is in this context that ETS calls must be given a high probability=20
>of meeting the QoS criteria, even when network congestion and or damage

>are severe.

I wonder if we are communicating here. RFC 3246 is designed to operate
in a congested network, and to operate in a network in which routing may
change. The comment in the mlpp-that-works draft - which if it is giving
cause for confusion I will happily remove - is basically me bending over
backwards to be intellectually honest and see if I can find a case where
CAC may be insufficient.

[jpg] And I was blatantly taking advantage of your "bending over
backwards"
- to identify a case where CAC may not be sufficient for the ETS Voice
application.

Think about this for a moment: the argument being placed for MLEF is
that even in the presence of some loss, voice is intelligible. If that
is true, then just after a route change, during the brief interval
between the data flow being switched to a new path and the CAC following
that and either OKing the flow or deciding to shut down the call, when
there may be some loss, voice should be equally intelligible, no?

 Or does the argument that voice is able to work around a modest level
of loss only work when it comes from the proponents of multiple code
points?

[jpg] In the ETS voice scenario, I am NOT trying to say that "voice is
able to work around a modest level of loss".  I am taking the highly
elitist position that, if there is a situation where SOME "calls" are
going to have to lose packets, I want it to be the non-ETS calls
(whether or not the non-ETS calls remain intelligible or not) that lose
packets, and not the ETS calls.

[jpg] I further agree that, in any situation where "calls" are losing
packets, it is highly desirable to "do something" so that you have some
calls that remain up and intelligible, and so that other calls go away,
rather than using up bandwidth sending packets that are not "useful"
from the end user's perspective.  And whatever that "something" is, I
want it to be able (assuming the appropriate business agreements are in
place) to slant the odds in favor of the ETS calls being the ones that
stay up and intelligible.


Janet
<< Attachment Removed : C.DTF >>




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From: Janet Gunn <jgunn@ix.netcom.com>
Reply-To: Janet Gunn <jgunn@ix.netcom.com>
To: steve.norreys@bt.com, jgunn6@csc.com
Subject: RE: References to MLPP that works Re: [Ieprep] I-D
 ACTION:draft-ietf-ieprep-domain-req-00.txt
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Steve,

I completely agree that pre-emption has no role in ETS voice applications.

In addition to (or perhaps in part because of) your points, anything resembling pre-emption on a service connected with the US PSTN is a political/legal non-starter.  Even if the governement wanted it (which they don't) the provider's lawyers would soon put a stop to it.

I was/am simply trying to extract anything that may be useful in the voice ETS domain from the discussion of MLPP over IP.

HOWEVER-
In a circuit switched network, if a link or node (switch) goes down, all calls using the node or link are lost.  

But in an IP network, if a link or switch goes down, it is possible to reroute the calls on the fly.  If there is sufficient capacity, all is well, and all calls remain up, and intelligible.  But if there is insufficient capacity available after rerouting, you have  (at least) 3 choices 

1) leave all the calls up, each one will have a significant drop rate, resulting in poor intelligibility.  Hope that some of the calls will hang up and try again, leaving additional capacity for the calls that didn't hang up (the most stubborn users win).
2) randomly "shut down" affected calls, until the remaining calls have enough capacity to be intelligible.
3) use some sort of priority scheme to determine which calls to shut down and which to leave up.

I don't think I would call any of those three options "pre-emption".

Janet

-----Original Message-----
From: steve.norreys@bt.com
Sent: Feb 3, 2004 10:42 AM
To: jgunn6@csc.com
Cc: ieprep@ietf.org
Subject: RE: References to MLPP that works Re: [Ieprep] I-D ACTION:draft-ietf-ieprep-domain-req-00.txt

Janet and all

I have been biting my tongue (ops fingers) for some time now with
regards to the MLPP service for supporting ETS in IP networks. As this
service has a potential problem as shown by the following scenario:

If a call/session is in progress between a non authorised ETS user and a
emergency service then this should never be subject to pre-emption by a
authorised ETS user, it could have fatal consequences. This by extension
could be applied to a call/session in progress between non authorised
ETS users, as this contact could be the only point of hope for someone
in trouble.

In other words the only justification for a pre-emption by a authorised
ETS user is that it is only applied to call/session if both the content
and context have been analysed. This seems to be a lot of work in a
congested network.

I have no problems with multiple precedence's or priorities if they are
required but I only see the need for two a 'normal' call/session and a
'ETS' call/session. 

This shows that there maybe a need to take into account the consequences
of the requirement to support MLPP especially in regard to pre-emption. 

Regards

Steve Norreys  
Signalling and Protocols
Tel:       +441277323220
email    steve.norreys@bt.com


-----Original Message-----
From: ieprep-admin@ietf.org [mailto:ieprep-admin@ietf.org] On Behalf Of
Janet P Gunn
Sent: 02 February 2004 22:13
To: Fred Baker
Cc: ieprep@ietf.org
Subject: References to MLPP that works Re: [Ieprep] I-D
ACTION:draft-ietf-ieprep-domain-req-00.txt



Comments in line.


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

This is a PRIVATE message. If you are not the intended recipient, please
delete without copying and kindly advise us by e-mail of the mistake in
delivery. NOTE: Regardless of content, this e-mail shall not operate to
bind CSC to any order or other contract unless pursuant to explicit
written agreement or government initiative expressly permitting the use
of e-mail for such purpose.
------------------------------------------------------------------------
----------------




 

                      Fred Baker <fred

                      @cisco.com>              To:      Janet P Gunn
<jgunn6@csc.com>                                            
                                               cc:      ieprep@ietf.org

                      01/27/2004 09:35         Subject: Re: [Ieprep] I-D
ACTION:draft-ietf-ieprep-domain-req-00.txt              
                      PM

 

 





At 12:32 PM 1/27/2004, Janet P Gunn wrote:
>As Fred Baker says in draft-baker-tsvwg-mlpp-that-works.00.txt, if call

>admission is working properly, traffic marking is not needed in normal 
>operations.

lest I be misunderstood, let me restate that. What I wrote was:

    One will ask the value of the multiple DSCPs. They are, in fact, of
    limited to no value in normal operation, as all vice and video
    traffic should have been admitted, and therefore capacity will have
    been assigned to them. The behavior of this service will be
    indistinguishable from the EF PHB regardless of traffic marking.

In retrospect, I wonder what the vice traffic would be :^)

[jpg] Maybe all that "organ enlargement" spam?

This is not to be understood as saying "no traffic marking is required";
one still needs RFC 3246. But if bandwidth admission is operating
correctly, having N different marks will be operationally
indistinguishable from having a single EF mark, as the total number of
bytes on the wire will be the same. jpg], I did understand that.  I was
oversimplifying to make my point.

>It is in this context that ETS calls must be given a high probability 
>of meeting the QoS criteria, even when network congestion and or damage

>are severe.

I wonder if we are communicating here. RFC 3246 is designed to operate
in a congested network, and to operate in a network in which routing may
change. The comment in the mlpp-that-works draft - which if it is giving
cause for confusion I will happily remove - is basically me bending over
backwards to be intellectually honest and see if I can find a case where
CAC may be insufficient.

[jpg] And I was blatantly taking advantage of your "bending over
backwards"
- to identify a case where CAC may not be sufficient for the ETS Voice
application.

Think about this for a moment: the argument being placed for MLEF is
that even in the presence of some loss, voice is intelligible. If that
is true, then just after a route change, during the brief interval
between the data flow being switched to a new path and the CAC following
that and either OKing the flow or deciding to shut down the call, when
there may be some loss, voice should be equally intelligible, no?

 Or does the argument that voice is able to work around a modest level
of loss only work when it comes from the proponents of multiple code
points?

[jpg] In the ETS voice scenario, I am NOT trying to say that "voice is
able to work around a modest level of loss".  I am taking the highly
elitist position that, if there is a situation where SOME "calls" are
going to have to lose packets, I want it to be the non-ETS calls
(whether or not the non-ETS calls remain intelligible or not) that lose
packets, and not the ETS calls.

[jpg] I further agree that, in any situation where "calls" are losing
packets, it is highly desirable to "do something" so that you have some
calls that remain up and intelligible, and so that other calls go away,
rather than using up bandwidth sending packets that are not "useful"
from the end user's perspective.  And whatever that "something" is, I
want it to be able (assuming the appropriate business agreements are in
place) to slant the odds in favor of the ETS calls being the ones that
stay up and intelligible.


Janet
<< Attachment Removed : C.DTF >>




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Date: Tue, 03 Feb 2004 15:37:55 -0800
To: Janet P Gunn <jgunn6@csc.com>
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At 02:12 PM 2/2/2004, Janet P Gunn wrote:
>[jpg] In the ETS voice scenario, I am NOT trying to say that "voice is 
>able to work around a modest level of loss".  I am taking the highly 
>elitist position that, if there is a situation where SOME "calls" are 
>going to have to lose packets, I want it to be the non-ETS calls (whether 
>or not the non-ETS calls remain intelligible or not) that lose packets, 
>and not the ETS calls.

I'll agree that it is elitist :^)

However, I'll submit that if there is bandwidth-aware CAC, there should not 
be a great amount of loss. Loss will come from periods in which variable 
rate codecs (VAD etc) happen to all have an "up" moment and go over the 
top, resulting in a few packets lost from each call, or from the normal 
conditions (say "wireless") that cause random loss in networks. If there is 
not bandwidth-aware CAC, then we are discussing the depth of 
overprovisioning - either you did enough or you didn't. If you did, I'm not 
sure what we're discussing - there is going to be plenty of bandwidth, and 
the SIP reason code and resource priority stuff handles call collisions. If 
you didn't, we're not talking about occasional dropped packets, we're 
discussing lost packet groupings in rough multiples of individual call 
volumes. In the latter case, one really wants to refuse or preempt calls.

The discussion of what packets to drop when on alternate Tuesdays the moon 
skips a beat in its orbit seems a very secondary consideration to 
delivering a solid service all of the time.

>[jpg] I further agree that, in any situation where "calls" are losing 
>packets, it is highly desirable to "do something" so that you have some 
>calls that remain up and intelligible, and so that other calls go away, 
>rather than using up bandwidth sending packets that are not "useful" from 
>the end user's perspective.  And whatever that "something" is, I want it 
>to be able (assuming the appropriate business agreements are in place) to 
>slant the odds in favor of the ETS calls being the ones that stay up and 
>intelligible.

I don't know if you heard about the demo we showed DISA and others in 
December. I don't think anybody is arguing that the ETS/MLPP/whatever calls 
need to be intelligible. The issue is that the calls you keep are paying 
for their bandwidth too, and are not getting the service you sold them if 
you're heaving their packets willy-nilly. At the end of the day, depending 
only on packet dropping (and yes, Mike, I heard you say in private email 
that you like CAC too, but I'm looking for the architectural document that 
says that and I'm having trouble finding it) doesn't deliver that. 
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Hi, Scott & Kimberly,

I am writing a new internet-draft. And I want to present this draft 
in the next meeting in Korea. The abstract of the draft is written in 
below. 

Abstract

   This document presents a usage of anycast in support of Emergency 
   Telecommunications Services (ETS) within the context of IP telephony. 
   In the Public Switched Telephone Network (PSTN), 911 is reserved to 
   call to the closest designated emergency agency. Instead of 911, 
   this document presents using anycast addresses within IP architecture 
   and service models for in support of ETS.


Now I'm not sure if I will be able to submit this draft till next Monday.
If possible, please add my presentation to time schedule of the next meeting.
The presentation may take 10 minitus.

Regards,
Kosuke TOMONAGA


On Thu, 15 Jan 2004 09:46:25 -0500 (EST)
sob@harvard.edu (Scott Bradner) wrote:

> 
> at this point we do not plan to have an ieprep meeting in Korea 
> 
> if feel very strongly that we should please let us know
> 
> Scott & Kimberly
> 
> _______________________________________________
> Ieprep mailing list
> Ieprep@ietf.org
> https://www1.ietf.org/mailman/listinfo/ieprep
> 

_______________________________________________
Ieprep mailing list
Ieprep@ietf.org
https://www1.ietf.org/mailman/listinfo/ieprep



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Date: Wed, 4 Feb 2004 16:23:53 +0900
From: Kosuke TOMONAGA <tomonaga@ale.csce.kyushu-u.ac.jp>
To: sob@harvard.edu (Scott Bradner)
Cc: ieprep@ietf.org
Subject: Re: [Ieprep] IEPREP in Korea
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Sorry, I sended a wrong mail including japanese font.
I submit again.

Regards,
Kosuke TOMONAGA

---
Hi, Scott & Kimberly,

I am writing a new internet-draft. And I want to present this draft 
in the next meeting in Korea. The abstract of the draft is written in 
below. 

Abstract

   This document presents a usage of anycast in support of Emergency 
   Telecommunications Services (ETS) within the context of IP telephony. 
   In the Public Switched Telephone Network (PSTN), 911 is reserved to 
   call to the closest designated emergency agency. Instead of 911, 
   this document presents using anycast addresses within IP architecture 
   and service models for in support of ETS.


Now I'm not sure if I will be able to submit this draft till next Monday.
If possible, please add my presentation to time schedule of the next meeting.
The presentation may take 10 minitus.

Regards,
Kosuke TOMONAGA


On Thu, 15 Jan 2004 09:46:25 -0500 (EST)
sob@harvard.edu (Scott Bradner) wrote:

> 
> at this point we do not plan to have an ieprep meeting in Korea 
> 
> if feel very strongly that we should please let us know
> 
> Scott & Kimberly
> 
> _______________________________________________
> Ieprep mailing list
> Ieprep@ietf.org
> https://www1.ietf.org/mailman/listinfo/ieprep
> 

_______________________________________________
Ieprep mailing list
Ieprep@ietf.org
https://www1.ietf.org/mailman/listinfo/ieprep



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Date: Wed, 4 Feb 2004 20:36:35 +0900
From: Kosuke TOMONAGA <tomonaga@ale.csce.kyushu-u.ac.jp>
To: "King, Kimberly  S." <KIMBERLY.S.KING@saic.com>
Cc: ieprep@ietf.org
Subject: Re: [Ieprep] IEPREP in Korea
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Hello, Kimberly,

> Please submit your draft but we won't have an IEPREP meeting in Korea.

Thank you, I'll submit my draft. But I feel very strongly that we should 
hold the meeting. Because we did not have an IEPREP meeting in the previous 
IETF meeting.

Regards,
Kosuke TOMONAGA


On Wed, 4 Feb 2004 04:19:20 -0500 
"King, Kimberly  S." <KIMBERLY.S.KING@saic.com> wrote:

> Hello,
> 
> Please submit your draft but we won't have an IEPREP meeting in Korea.
> 
> Kimberly
> 
> -----Original Message-----
> From: ieprep-admin@ietf.org
> To: sob@harvard.edu
> Cc: ieprep@ietf.org
> Sent: 2/4/2004 2:23 AM
> Subject: Re: [Ieprep] IEPREP in Korea
> 
> Sorry, I sended a wrong mail including japanese font.
> I submit again.
> 
> Regards,
> Kosuke TOMONAGA
> 
> ---
> Hi, Scott & Kimberly,
> 
> I am writing a new internet-draft. And I want to present this draft 
> in the next meeting in Korea. The abstract of the draft is written in 
> below. 
> 
> Abstract
> 
>    This document presents a usage of anycast in support of Emergency 
>    Telecommunications Services (ETS) within the context of IP telephony.
> 
>    In the Public Switched Telephone Network (PSTN), 911 is reserved to 
>    call to the closest designated emergency agency. Instead of 911, 
>    this document presents using anycast addresses within IP architecture
> 
>    and service models for in support of ETS.
> 
> 
> Now I'm not sure if I will be able to submit this draft till next
> Monday.
> If possible, please add my presentation to time schedule of the next
> meeting.
> The presentation may take 10 minitus.
> 
> Regards,
> Kosuke TOMONAGA
> 
> 
> On Thu, 15 Jan 2004 09:46:25 -0500 (EST)
> sob@harvard.edu (Scott Bradner) wrote:
> 
> > 
> > at this point we do not plan to have an ieprep meeting in Korea 
> > 
> > if feel very strongly that we should please let us know
> > 
> > Scott & Kimberly
> > 
> > _______________________________________________
> > Ieprep mailing list
> > Ieprep@ietf.org
> > https://www1.ietf.org/mailman/listinfo/ieprep
> > 
> 
> _______________________________________________
> Ieprep mailing list
> Ieprep@ietf.org
> https://www1.ietf.org/mailman/listinfo/ieprep
> 

_______________________________________________
Ieprep mailing list
Ieprep@ietf.org
https://www1.ietf.org/mailman/listinfo/ieprep



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Subject: Re: [Ieprep] IEPREP in Korea
Cc: ieprep@ietf.org
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we do not plan to have an ieprep session during the meeting but even if we
did we do not provide time to present internet drafts that have not
had discussion on the list 

note that the pre meeting cutoff is 9am US eastern time on monday - 
any new ID submission received after that time will be rejected
(the cutoff for revised IDs is teh following monday)

Scott


---
From tomonaga@ale.csce.kyushu-u.ac.jp  Wed Feb  4 02:00:37 2004
X-Original-To: sob@newdev.harvard.edu
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From: =?ISO-2022-JP?B?GyRCRCtEORsoQiAbJEI5LzJwGyhC?= <tomonaga@ale.csce.kyushu-u.ac.jp>
To: sob@harvard.edu (Scott Bradner)
Cc: ieprep@ietf.org
Subject: Re: [Ieprep] IEPREP in Korea
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Hi, Scott & Kimberly,

I am writing a new internet-draft. And I want to present this draft 
in the next meeting in Korea. The abstract of the draft is written in 
below. 

Abstract

   This document presents a usage of anycast in support of Emergency 
   Telecommunications Services (ETS) within the context of IP telephony. 
   In the Public Switched Telephone Network (PSTN), 911 is reserved to 
   call to the closest designated emergency agency. Instead of 911, 
   this document presents using anycast addresses within IP architecture 
   and service models for in support of ETS.


Now I'm not sure if I will be able to submit this draft till next Monday.
If possible, please add my presentation to time schedule of the next meeting.
The presentation may take 10 minitus.

Regards,
Kosuke TOMONAGA


On Thu, 15 Jan 2004 09:46:25 -0500 (EST)
sob@harvard.edu (Scott Bradner) wrote:

> 
> at this point we do not plan to have an ieprep meeting in Korea 
> 
> if feel very strongly that we should please let us know
> 
> Scott & Kimberly
> 
> _______________________________________________
> Ieprep mailing list
> Ieprep@ietf.org
> https://www1.ietf.org/mailman/listinfo/ieprep
> 


_______________________________________________
Ieprep mailing list
Ieprep@ietf.org
https://www1.ietf.org/mailman/listinfo/ieprep



From exim@www1.ietf.org  Wed Feb  4 08:04:47 2004
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'?? ??'" <tomonaga@ale.csce.kyushu-u.ac.jp>
Cc: ieprep@ietf.org
Subject: RE: [Ieprep] IEPREP in Korea
Date: Wed, 4 Feb 2004 08:00:10 -0500 
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I think this draft is not appropriate for IEPREP, at least as I interpret
the draft.  I believe the IEPREP charter deals with how the Internet 
can be used in disasters, where existing facilities may be needed by 
responders to organize relief.

The normal call for help, for which 9-1-1 would be the phone number 
in North America, is out of scope.  We are dealing with these issues 
in sipping, and there is a draft (currently expired, but expected to be 
reissued shortly) that covers the specific problem raised by this draft 
(9-1-1 is not an internationally recognized directory number for 
emergencies), although it proposes a different solution.

Brian


> -----Original Message-----
> From: tomonaga@ale.csce.kyushu-u.ac.jp 
> [mailto:tomonaga@ale.csce.kyushu-u.ac.jp]
> Sent: Wednesday, February 04, 2004 2:12 AM
> To: sob@harvard.edu
> Cc: ieprep@ietf.org
> Subject: Re: [Ieprep] IEPREP in Korea
> 
> 
> Hi, Scott & Kimberly,
> 
> I am writing a new internet-draft. And I want to present this draft 
> in the next meeting in Korea. The abstract of the draft is written in 
> below. 
> 
> Abstract
> 
>    This document presents a usage of anycast in support of Emergency 
>    Telecommunications Services (ETS) within the context of IP 
> telephony. 
>    In the Public Switched Telephone Network (PSTN), 911 is 
> reserved to 
>    call to the closest designated emergency agency. Instead of 911, 
>    this document presents using anycast addresses within IP 
> architecture 
>    and service models for in support of ETS.
> 
> 
> Now I'm not sure if I will be able to submit this draft till 
> next Monday.
> If possible, please add my presentation to time schedule of 
> the next meeting.
> The presentation may take 10 minitus.
> 
> Regards,
> Kosuke TOMONAGA
> 
> 
> On Thu, 15 Jan 2004 09:46:25 -0500 (EST)
> sob@harvard.edu (Scott Bradner) wrote:
> 
> > 
> > at this point we do not plan to have an ieprep meeting in Korea 
> > 
> > if feel very strongly that we should please let us know
> > 
> > Scott & Kimberly
> > 
> > _______________________________________________
> > Ieprep mailing list
> > Ieprep@ietf.org
> > https://www1.ietf.org/mailman/listinfo/ieprep
> > 
> 
> _______________________________________________
> Ieprep mailing list
> Ieprep@ietf.org
> https://www1.ietf.org/mailman/listinfo/ieprep
> 

_______________________________________________
Ieprep mailing list
Ieprep@ietf.org
https://www1.ietf.org/mailman/listinfo/ieprep



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> The normal call for help, for which 9-1-1 would be the phone number 
> in North America, is out of scope.

Brian is corect that citizen-to-authority emergency calling is out of
scope for ieprep and is being addressed in sipping

Scott

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From: "Ken Carlberg" <carlberg@g11.org.uk>
To: <steve.norreys@bt.com>
Cc: <ieprep@ietf.org>
Subject: RE: References to MLPP that works Re: [Ieprep] I-D ACTION:draft-ietf-ieprep-domain-req-00.txt
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Steve,

Just a couple of comments for clarification to hopefully alleviate some
of your concerns...

> If a call/session is in progress between a non authorised ETS user and
a
> emergency service then this should never be subject to pre-emption by
a
> authorised ETS user, it could have fatal consequences. This by
extension
> could be applied to a call/session in progress between non authorised
> ETS users, as this contact could be the only point of hope for someone
> in trouble.

In the soon-to-be-released RFC-3689, General Requirements for ETS, there
are a couple of pertinent excerpts about "policy".  Paraphrased, it says
that a set of labels may exist and that they may have a stated
relationship to each other.  In the case of MLPP, this would include
preemption. HOWEVER, that relationship is only for *that* set.  This
means that the characteristic of preemption does not automatically
correlate to a different set of defined labels.  Instead, that
relationship is defined by local policy.

Put another way, the fact that MLPP has precedence does not mean that
the same characteristic automatically applies to other labels (ie, other
ETS type labels) or non labeled traffic.  Instead, you the operator (if
you support ETS labels) decide the relationship between different sets
of ETS type labels, as well as the relationship between ETS and non-ETS
labeled traffic.  If this is still not clear, contact me privately and I
will expand.

Note: In the case of BT, Oftel may make that decision for you based on
laws/regulations, but that's outside the scope of the technology.  In
the case of the US PSTN, automated preemption is not allowed.  

-ken

ps, the predominance of MLPP systems in use today are in private
(non-PSTN) systems.


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--NextPart


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


        RFC 3689

        Title:      General Requirements for
                    Emergency Telecommunication Service (ETS)
        Author(s):  K. Carlberg, R. Atkinson
        Status:     Informational
        Date:       February 2004
        Mailbox:    k.carlberg@cs.ucl.ac.uk, rja@extremenetworks.com
        Pages:      10
        Characters: 21680
        Updates/Obsoletes/SeeAlso:    None

        I-D Tag:    draft-ietf-ieprep-ets-general-04.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3689.txt


This document presents a list of general requirements in support of
Emergency Telecommunications Service (ETS).  Solutions to these
requirements are not presented in this document.  Additional
requirements pertaining to specific applications, or types of
applications, are to be specified in separate document(s).

This document is a product of the Internet Emergency Preparedness
Working Group of the IETF.

This memo provides information for the Internet community.  It does
not specify an Internet standard of any kind.  Distribution of this
memo is unlimited.

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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

...

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--NextPart


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


        RFC 3690

        Title:      IP Telephony Requirements for
                    Emergency Telecommunication Service (ETS)
        Author(s):  K. Carlberg, R. Atkinson
        Status:     Informational
        Date:       February 2004
        Mailbox:    k.carlberg@cs.ucl.ac.uk, rja@extremenetworks.com
        Pages:      7
        Characters: 13919
        Updates/Obsoletes/SeeAlso:    None

        I-D Tag:    draft-ietf-ieprep-ets-telephony-07.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3690.txt


This document presents a list of requirements in support of Emergency
Telecommunications Service (ETS) within the context of IP
telephony.  It is an extension to the general requirements presented
in RFC 3689.  Solutions to these requirements are not presented in
this document.

This document is a product of the Internet Emergency Preparedness
Working Group of the IETF.

This memo provides information for the Internet community.  It does
not specify an Internet standard of any kind.  Distribution of this
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Authors, for further information.


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

...

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--NextPart

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

	Title		: Framework for Supporting IEPS in IP Telephony
	Author(s)	: K. Carlberg, I. Brown, C. Beard
	Filename	: draft-ietf-ieprep-framework-08.txt
	Pages		: 27
	Date		: 2004-2-9
	
This document presents a framework for supporting authorized
emergency related communication within the context of IP telephony.
We present a series of objectives that reflect a general view of how
authorized emergency service, in line with the Emergency
Telecommunications Service (ETS), should be realized within today's
IP architecture and service models.  From these objectives, we
present a corresponding set of protocols and capabilities, which
provide a more specific set of recommendations regarding existing
IETF protocols.  Finally, we present two scenarios that act as
guiding models for the objectives and functions listed in this
document.  These, models, coupled with an example of an existing
service in the PSTN, contribute to a constrained solution space.

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From exim@www1.ietf.org  Mon Feb  9 17:26:10 2004
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From: Janet Gunn <jgunn@ix.netcom.com>
Reply-To: Janet Gunn <jgunn@ix.netcom.com>
To: Scott Bradner <sob@harvard.edu>, jgunn6@csc.com, jmpolk@cisco.com
Subject: Re: CAC RE: [Ieprep] I-D ACTION:draft-ietf-ieprep-domain-req-00.txt
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I have finally had a chance to do some more reading, and I think I fianlly understand the source of my confusion (and probably some others).

I had assumed that the term "Call Admission Control" was active only only at the START of a call- that once a call had been admitted in the first place, there was no further CAC involvement.  

However, reading between the lines in several posts, and relating that to some of the reference documents, it appears that the term CAC is being applied to a mechanism which remains involved throught the live of the call, and can "rescind" the admission during the course of the call, if necessary.

Is this right, or am I just barking up a different wrong tree?

Janet

-----Original Message-----
From: Scott Bradner <sob@harvard.edu>
Sent: Feb 3, 2004 5:31 AM
To: jgunn6@csc.com, jmpolk@cisco.com
Cc: fred@cisco.com, ieprep@ietf.org, steves@shentel.net
Subject: Re: CAC RE: [Ieprep] I-D ACTION:draft-ietf-ieprep-domain-req-00.txt

> [jpg] It is not clear to me how CAC says "sorry, but there was another call
> with higher priority, and I have to shut your call down now" once the call
> has been established.  I was under the impression that, once the call was
> established, the router did not distinguish between individual calls.  I
> would be glad to be enlightened on that point.

note that the router is not involved in establishing the call - its all
packets to the router

but if the call QoS is maintained by RSVP (as Fred suggested) there is 
less of an issue - a router can reject RSVP sessions even after they
have started

Scott

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At 02:20 PM 2/9/2004 -0800, Janet Gunn wrote:
>I have finally had a chance to do some more reading, and I think I fianlly 
>understand the source of my confusion (and probably some others).
>
>I had assumed that the term "Call Admission Control" was active only only 
>at the START of a call- that once a call had been admitted in the first 
>place, there was no further CAC involvement.
>
>However, reading between the lines in several posts, and relating that to 
>some of the reference documents, it appears that the term CAC is being 
>applied to a mechanism which remains involved throught the live of the 
>call, and can "rescind" the admission during the course of the call, if 
>necessary.

What you were believing was the case is called "Locations based CAC" (done 
once at start-up, and not again). This is a hack, but kinda sorta works in 
very limited links between 2 locations (not daisy-chained mind you, the 
complexity and topology knowledge is just not really possible then).

What you have realized about RSVP is that the CAC functionality is from 
call inception, makes sure there is enough resources to complete a call at 
a known bandwidth (errors the set-up if there isn't enough), and the 
routers in between remain in a "soft state" condition for all the RSVP 
"flows" that have been accepted through that interface for the duration of 
all the flows (or calls) until their end, which is either by a normal end 
of a call (user hang-up) or a preemption event at a congested router 
interface (to make room for another flow/call).


>Is this right, or am I just barking up a different wrong tree?

the crowd applauds!!


>Janet
>
>-----Original Message-----
>From: Scott Bradner <sob@harvard.edu>
>Sent: Feb 3, 2004 5:31 AM
>To: jgunn6@csc.com, jmpolk@cisco.com
>Cc: fred@cisco.com, ieprep@ietf.org, steves@shentel.net
>Subject: Re: CAC RE: [Ieprep] I-D ACTION:draft-ietf-ieprep-domain-req-00.txt
>
> > [jpg] It is not clear to me how CAC says "sorry, but there was another call
> > with higher priority, and I have to shut your call down now" once the call
> > has been established.  I was under the impression that, once the call was
> > established, the router did not distinguish between individual calls.  I
> > would be glad to be enlightened on that point.
>
>note that the router is not involved in establishing the call - its all
>packets to the router
>
>but if the call QoS is maintained by RSVP (as Fred suggested) there is
>less of an issue - a router can reject RSVP sessions even after they
>have started
>
>Scott
>
>_______________________________________________
>Ieprep mailing list
>Ieprep@ietf.org
>https://www1.ietf.org/mailman/listinfo/ieprep


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


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Subject: Re: CAC RE: [Ieprep] I-D ACTION:draft-ietf-ieprep-domain-req-00.txt
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In a message dated 2/9/2004 5:23:37 PM Eastern Standard Time, 
jgunn@ix.netcom.com writes:


> I had assumed that the term "Call Admission Control" was active only only 
> at the START of a call- that once a call had been admitted in the first place, 
> there was no further CAC involvement.  
> 
> However, reading between the lines in several posts, and relating that to 
> some of the reference documents, it appears that the term CAC is being applied 
> to a mechanism which remains involved throught the live of the call, and can 
> "rescind" the admission during the course of the call, if necessary.
> 


This is a term which needs consistent definition before we continue. It's 
being used in different ways, including as you noted above.

Since the term is "Admission", I believe it only makes sense to use "CAC" it 
to refer to the control that is placed on deciding whether or not to admit a 
new call (i.e., packet flow), not on anything that is done later to decide 
whether to preempt a call or terminate a packet flow.

Mike


--part1_11c.2b203b73.2d597ceb_boundary
Content-Type: text/html; charset="US-ASCII"
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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 2/9/2004 5:23:37 PM Eastern Standard Time, jgunn@ix.netcom.com writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">I had assumed that the term=
 "Call Admission Control" was active only only at the START of a call- that=20=
once a call had been admitted in the first place, there was no further CAC i=
nvolvement. &nbsp;
<BR>
<BR>However, reading between the lines in several posts, and relating that t=
o some of the reference documents, it appears that the term CAC is being app=
lied to a mechanism which remains involved throught the live of the call, an=
d can "rescind" the admission during the course of the call, if necessary.
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">
<BR>
<BR>This is a term which needs consistent definition before we continue. It'=
s being used in different ways, including as you noted above.
<BR>
<BR>Since the term is "Admission", I believe it only makes sense to use "CAC=
" it to refer to the control that is placed on deciding whether or not to ad=
mit a new call (i.e., packet flow), not on anything that is done later to de=
cide whether to preempt a call or terminate a packet flow.
<BR>
<BR>Mike
<BR></FONT></HTML>

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Subject: Re: CAC RE: [Ieprep] I-D
  ACTION:draft-ietf-ieprep-domain-req-00.txt
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I'm sorry Mike, but that is the functionality RSVP does and has done for 
many many years.... I'm not going to change it. Further, the maintenance 
CAC function (what you would call the part after initial admission) is what 
RSVP does. When referring to CAC and RSVP, this is the definition that is 
used in the IETF for voice and video flows.

When in Rome, you do what the Roman's do....

At 07:16 PM 2/9/2004 -0500, Mpierce1@aol.com wrote:
>In a message dated 2/9/2004 5:23:37 PM Eastern Standard Time, 
>jgunn@ix.netcom.com writes:
>
>
>>I had assumed that the term "Call Admission Control" was active only only 
>>at the START of a call- that once a call had been admitted in the first 
>>place, there was no further CAC involvement.
>>
>>However, reading between the lines in several posts, and relating that to 
>>some of the reference documents, it appears that the term CAC is being 
>>applied to a mechanism which remains involved throught the live of the 
>>call, and can "rescind" the admission during the course of the call, if 
>>necessary.
>
>
>
>This is a term which needs consistent definition before we continue. It's 
>being used in different ways, including as you noted above.
>
>Since the term is "Admission", I believe it only makes sense to use "CAC" 
>it to refer to the control that is placed on deciding whether or not to 
>admit a new call (i.e., packet flow), not on anything that is done later 
>to decide whether to preempt a call or terminate a packet flow.
>
>Mike


cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


_______________________________________________
Ieprep mailing list
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https://www1.ietf.org/mailman/listinfo/ieprep



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At 04:16 PM 2/9/2004, Mpierce1@aol.com wrote:
>Since the term is "Admission", I believe it only makes sense to use "CAC" 
>it to refer to the control that is placed on deciding whether or not to 
>admit a new call (i.e., packet flow), not on anything that is done later 
>to decide whether to preempt a call or terminate a packet flow.

perhaps.

As James noted, there are a variety of procedures that are called CAC: 
these are generally call counters in SIP Proxies or H.323 Gatekeepers, and 
MOSAIC defines something based on Bandwidth Broker research. These, as you 
say, these procedures look at the call when it is being negotiated and 
never look at it again.

RSVP, as defined in RFC 2205, is continuous. It maintains a continuous 
knowledge of what sessions are up and what sessions are going away, 
reinstalls them if the routing path changes, and is quite capable of 
preempting a call if need be. This is discussed in some detail in
     http://www.ietf.org/internet-drafts/draft-baker-tsvwg-mlpp-that-works-01.txt

which will be posted later on this week. The -00 version touches on it, but 
not in as great detail.

To assert that a bandwidth-and-routing aware protocol such as RSVP is not 
"call admission control" does some violence to the English language, IMHO. 
If it is not admitting calls, I'm not sure what it is doing. Yes, in 
addition is performs preemption; to RSVP, those are different outcomes of 
the same process.

I have a list of the relevant IETF standards at
     ftp://ftpeng.cisco.com/fred/mlpp/rsvp.html

if you're interested, you might want to review how preemption works in RSVP
      ftp://ftp.rfc-editor.org/in-notes/rfc3181.txt 
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In a message dated 2/9/2004 8:59:41 PM Eastern Standard Time, fred@cisco.com 
writes:


> As James noted, there are a variety of procedures that are called CAC: 
> these are generally call counters in SIP Proxies or H.323 Gatekeepers, and 
> MOSAIC defines something based on Bandwidth Broker research. These, as you 
> say, these procedures look at the call when it is being negotiated and 
> never look at it again.
> 
> RSVP, as defined in RFC 2205, is continuous. It maintains a continuous 
> knowledge of what sessions are up and what sessions are going away, 
> reinstalls them if the routing path changes, and is quite capable of 
> preempting a call if need be. This is discussed in some detail in
>      
> http://www.ietf.org/internet-drafts/draft-baker-tsvwg-mlpp-that-works-01.txt
> 
> which will be posted later on this week. The -00 version touches on it, but 
> not in as great detail.
> 
> To assert that a bandwidth-and-routing aware protocol such as RSVP is not 
> "call admission control" does some violence to the English language, IMHO. 
> If it is not admitting calls, I'm not sure what it is doing. Yes, in 
> addition is performs preemption; to RSVP, those are different outcomes of 
> the same process.
> 



To all,

The issue that Janet raised and that I responded to had nothing to do with 
the functionality of RSVP. I am well aware that RSVP includes the notion of 
preempting existing flows, and I think we can make good use of that. I did not 
assert that RSVP is not "call admisison control". I said nothing about RSVP.

The issue raised by Janet is simply the definition of the term "Call 
Admission Control" or "CAC" and whether or not the term includes "preemption".

Since you're mentioned RSVP, looking through existing RSVP RFCs, there are 
references to "Admission Control" and RFC3181 mentions "Capacity Admission 
Control (CAC)". (Of course, RSVP does not refer to "Call Admission Control".)

RFC 2205 defines:

   o    Traffic control

        The entire set of machinery in the node that supplies requested
        QoS to data streams.  Traffic control includes packet
        classifier, packet scheduler, and admission control functions.

   o    Admission control

        A traffic control function that decides whether the packet
        scheduler in the node can supply the requested QoS while
        continuing to provide the QoS requested by previously-admitted
        requests.  See also "policy control" and "traffic control".

I would claim that the above definition of "Admission Control" does not 
include the function of preemption, although RSVP does include preemption.

I would suggest that as we talk more about preemption (of calls or packet 
flows), we need better definitions. It seems that "Traffic Control" should be 
extended to include a fourth component: preemption of packet flows. The 
definition of "Admisison Control" would remain as it is above in RFC 2205. As noted, 
CAC may cause preemption to occur, but I think it's better to define them as 
separate functions (and terms).

Mike Pierce


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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 2/9/2004 8:59:41 PM Eastern Standard Time, fred@cisco.com writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">As James noted, there are a=
 variety of procedures that are called CAC:=20
<BR>these are generally call counters in SIP Proxies or H.323 Gatekeepers, a=
nd=20
<BR>MOSAIC defines something based on Bandwidth Broker research. These, as y=
ou=20
<BR>say, these procedures look at the call when it is being negotiated and=20
<BR>never look at it again.
<BR>
<BR>RSVP, as defined in RFC 2205, is continuous. It maintains a continuous=20
<BR>knowledge of what sessions are up and what sessions are going away,=20
<BR>reinstalls them if the routing path changes, and is quite capable of=20
<BR>preempting a call if need be. This is discussed in some detail in
<BR> &nbsp;&nbsp;&nbsp;&nbsp;http://www.ietf.org/internet-drafts/draft-baker=
-tsvwg-mlpp-that-works-01.txt
<BR>
<BR>which will be posted later on this week. The -00 version touches on it,=20=
but=20
<BR>not in as great detail.
<BR>
<BR>To assert that a bandwidth-and-routing aware protocol such as RSVP is no=
t=20
<BR>"call admission control" does some violence to the English language, IMH=
O.=20
<BR>If it is not admitting calls, I'm not sure what it is doing. Yes, in=20
<BR>addition is performs preemption; to RSVP, those are different outcomes o=
f=20
<BR>the same process.
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">
<BR>
<BR>
<BR>To all,
<BR>
<BR>The issue that Janet raised and that I responded to had nothing to do wi=
th the functionality of RSVP. I am well aware that RSVP includes the notion=20=
of preempting existing flows, and I think we can make good use of that. I di=
d not assert that RSVP is not "call admisison control". I said nothing about=
 RSVP.
<BR>
<BR>The issue raised by Janet is simply the definition of the term "Call Adm=
ission Control" or "CAC" and whether or not the term includes "preemption".
<BR>
<BR>Since you're mentioned RSVP, looking through existing RSVP RFCs, there a=
re references to "Admission Control" and RFC3181 mentions "Capacity Admissio=
n Control (CAC)". (Of course, RSVP does not refer to "Call Admission Control=
".)
<BR>
<BR>RFC 2205 defines:
<BR>
<BR> &nbsp;&nbsp;o &nbsp;&nbsp;&nbsp;Traffic control
<BR>
<BR> &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;The entire set of machinery i=
n the node that supplies requested
<BR> &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;QoS to data streams. &nbsp;Tr=
affic control includes packet
<BR> &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;classifier, packet scheduler,=
 and admission control functions.
<BR>
<BR> &nbsp;&nbsp;o &nbsp;&nbsp;&nbsp;Admission control
<BR>
<BR> &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;A traffic control function th=
at decides whether the packet
<BR> &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;scheduler in the node can sup=
ply the requested QoS while
<BR> &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;continuing to provide the QoS=
 requested by previously-admitted
<BR> &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;requests. &nbsp;See also "pol=
icy control" and "traffic control".
<BR>
<BR>I would claim that the above definition of "Admission Control" does not=20=
include the function of preemption, although RSVP does include preemption.
<BR>
<BR>I would suggest that as we talk more about preemption (of calls or packe=
t flows), we need better definitions. It seems that "Traffic Control" should=
 be extended to include a fourth component: preemption of packet flows. The=20=
definition of "Admisison Control" would remain as it is above in RFC 2205. A=
s noted, CAC may cause preemption to occur, but I think it's better to defin=
e them as separate functions (and terms).
<BR>
<BR>Mike Pierce
<BR></FONT></HTML>

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In a message dated 2/10/2004 11:56:45 AM Eastern Standard Time, 
jgunn6@csc.com writes:


> Yes, I think it would be worthwhile to have a "definition" of CAC as a
> stand-alone posting on the list.   But I don't think we need to have a
> "discussion" about it.
> 

Well, it's hard to have a "definition" without some discussion, since it 
should be something that is agreed.

I think our "discussion" and the need is for a definition of "Call Admission 
Control", which I had used in my draft on preferential treatement examples, 
with the belief that there was a uniform understanding of what it meant. I 
believe it is an old term from the telephony environment, and what we are talking 
about here is "telephony". The application layer functions (call setup) haven't 
changed.

We should clearly separate the transport layer (e.g., RSVP, packet flows) 
from the application layer (call setup, preemption, release) with our terms. 
Unfortunately, RFC3181 uses the term "Capacity Admission Control (CAC)" in 
referring to packet transport.

Fred and James' comments related to RSVP, but RSVP never talked about 
"calls". As RSVP defines "Admission Control", I don't see the term
including "preemption", although RSVP clearly does.

In a message dated 2/10/2004 3:28:34 PM Eastern Standard Time, 
jmpolk@cisco.com writes:


>  define CAC in IP as:
> 
>          - set-up of the call at the endpoints (SIP does this), and
> 
>          - the ongoing monitoring process to keep the flow "functional"
> 
> this second bullet deals with packet admission, per packet to meet an 
> agreed upon bandwidth guarantee promised to all other nodes for that flow.
> 

Yes, I think this is fine, except that the first bullet is the application 
layer (call) and the second is the packet transport layer. So "CAC" would then 
be "Call/Capacity Admission Control".

CAC affects the initial setup of the call as well as the continuation of the 
call (packet flow). That's fine. I just think that the specific function of 
preemption of one call (packet flow) by another should be called something else 
for sake of clarity in descriptions, etc.

Mike Pierce


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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 2/10/2004 11:56:45 AM Eastern Standard Time, jgunn6@csc.com writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">Yes, I think it would be wo=
rthwhile to have a "definition" of CAC as a
<BR>stand-alone posting on the list. &nbsp;&nbsp;But I don't think we need t=
o have a
<BR>"discussion" about it.
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">
<BR>Well, it's hard to have a "definition" without some discussion, since it=
 should be something that is agreed.
<BR>
<BR>I think our "discussion" and the need is for a definition of "Call Admis=
sion Control", which I had used in my draft on preferential treatement examp=
les, with the belief that there was a uniform understanding of what it meant=
. I believe it is an old term from the telephony environment, and what we ar=
e talking about here is "telephony". The application layer functions (call s=
etup) haven't changed.
<BR>
<BR>We should clearly separate the transport layer (e.g., RSVP, packet flows=
) from the application layer (call setup, preemption, release) with our term=
s. Unfortunately, RFC3181 uses the term "Capacity Admission Control (CAC)" i=
n referring to packet transport.
<BR>
<BR>Fred and James' comments related to RSVP, but RSVP never talked about "c=
alls". As RSVP defines "Admission Control", I don't see the term
<BR>including "preemption", although RSVP clearly does.
<BR>
<BR>In a message dated 2/10/2004 3:28:34 PM Eastern Standard Time, jmpolk@ci=
sco.com writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px"> define CAC in IP as:
<BR>
<BR> &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;- set-up of the call at=
 the endpoints (SIP does this), and
<BR>
<BR> &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;- the ongoing monitorin=
g process to keep the flow "functional"
<BR>
<BR>this second bullet deals with packet admission, per packet to meet an=20
<BR>agreed upon bandwidth guarantee promised to all other nodes for that flo=
w.
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D3 PTSIZE=3D12 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0"></BLOCKQUOTE>
<BR></FONT><FONT  COLOR=3D"#000000" BACK=3D"#ffffff" style=3D"BACKGROUND-COL=
OR: #ffffff" SIZE=3D2 PTSIZE=3D10 FAMILY=3D"SANSSERIF" FACE=3D"Arial" LANG=
=3D"0">
<BR>Yes, I think this is fine, except that the first bullet is the applicati=
on layer (call) and the second is the packet transport layer. So "CAC" would=
 then be "Call/Capacity Admission Control".
<BR>
<BR>CAC affects the initial setup of the call as well as the continuation of=
 the call (packet flow). That's fine. I just think that the specific functio=
n of preemption of one call (packet flow) by another should be called someth=
ing else for sake of clarity in descriptions, etc.
<BR>
<BR>Mike Pierce
<BR></FONT></HTML>

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From: "Ken Carlberg" <carlberg@g11.org.uk>
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From: Internet-Drafts@ietf.org
Reply-to: Internet-Drafts@ietf.org
Date: Mon, 09 Feb 2004 16:21:19 -0500
Subject: [Ieprep] I-D ACTION:draft-ietf-ieprep-framework-08.txt
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This latest version includes changes based on private comments from Ran
Atkinson (primarily in the area of security), and Yakov Shafranovich.
My understanding is that ieprep chairs will soon announce a working
group last call on the draft with the hope of putting this puppy to
rest.

Cheers,

-ken

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

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

	Title		: Framework for Supporting IEPS in IP Telephony
	Author(s)	: K. Carlberg, I. Brown, C. Beard
	Filename	: 
	Pages		: 27
	Date		: 2004-2-9
	
This document presents a framework for supporting authorized
emergency related communication within the context of IP telephony.
We present a series of objectives that reflect a general view of how
authorized emergency service, in line with the Emergency
Telecommunications Service (ETS), should be realized within today's
IP architecture and service models.  From these objectives, we
present a corresponding set of protocols and capabilities, which
provide a more specific set of recommendations regarding existing
IETF protocols.  Finally, we present two scenarios that act as
guiding models for the objectives and functions listed in this
document.  These, models, coupled with an example of an existing
service in the PSTN, contribute to a constrained solution space.


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