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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Rex Buddenberg'" <budden@nps.navy.mil>, Mpierce1@aol.com
Cc: jmpolk@cisco.com, steves@shentel.net, jgunn@ix.netcom.com,
        carlberg@g11.org.uk, ieprep@ietf.org
Subject: RE: [Ieprep] RE:MLPPoIP - Preemption of low priority calls
Date: Mon, 1 Dec 2003 14:06:01 -0500 
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This discussion relates to another thread happening on dccp.
The basis of that thread is a discussion of whether it is possible
to design a protocol that can be used for real time data that is
"fair" with respect to TCP.  One of the opinions on this thread
(mine among others) is that fairness can only be achieved if TCP 
also changes to be aware of "constant bit rate"-like streams.  
If there was a mechanism introduced into TCP that worked to allow 
real time media and TCP to better co-exist, as well as protocols 
that allowed media streams to better co-exist with TCP, then perhaps 
there would be opportunities to address some of the concerns Rex raises.

Please also let me comment on:
> 	- cut back a VTC to voice only.  The psychologists tell 
> us that a
> talking-head VTC has visual cues that we use for about 5-10 minutes;
> after that we might as well turn the video off and just leave the
> audio.  
I'd like to see the citation on this research.  Our research shows that
people in the same room with one another look intently on the faces of
those talking, typically 90% of the time if engaged.  On a well designed
"VTC", you can observe the same effect, even an hour or more into a call.  
On a poorer quality system, participants tend to look around the room a 
lot.  I wonder if the research you refer to is out of date.

And yes Scott, a router can't be expected to tell the difference between
a well designed and a poorly designed video system. 

Brian

> -----Original Message-----
> From: Rex Buddenberg [mailto:budden@nps.navy.mil]
> Sent: Wednesday, November 26, 2003 4:47 PM
> To: Mpierce1@aol.com
> Cc: jmpolk@cisco.com; steves@shentel.net; jgunn@ix.netcom.com;
> carlberg@g11.org.uk; ieprep@ietf.org
> Subject: Re: [Ieprep] RE:MLPPoIP - Preemption of low priority calls
> 
> 
> On Wed, 2003-11-26 at 12:17, Mpierce1@aol.com wrote:
> ...
> > 
> > The current GSCR was written for circuit mode. Unless we get it
> > changed, it will apply to packet mode (even when it doesn't always
> > make sense). This is one of the areas we are trying to change, so it
> > correctly takes into account the advantages of using packet. 
> ...
> 
> 
> This whole conversation appears to be predicated on some 
> tacit and false
> assumptions.
> 
> The principal assumption seems to be that the entire capacity of a hop
> is taken up with VOICE conversations so that the solution to
> oversubscription is to shed some of those VOICE conversations.  
> 
> If you come to the conversation with a bellhead mentality, then we can
> probably read up on Erlangs and reach much the same 
> conclusion that the
> circuit switch world reached years ago.
> 
> 
> If you have a priority voice conversation and experience 
> congestion, it
> would seem to me that we would want to bump something other than other
> voice conversations:
> 	- anything file transfer (incl e-mail enclosures) with 
> .ppt in the file
> label.  The content:bit-bloat ratio there is terrible to the 
> extent that
> I'm calling powerpoint a virus these days.
> 	- cut back a VTC to voice only.  The psychologists tell 
> us that a
> talking-head VTC has visual cues that we use for about 5-10 minutes;
> after that we might as well turn the video off and just leave the
> audio.  The problem here is that we can't know by looking at port
> numbers whether someone has turned the camera around and is 
> sharing view
> of a crime scene
> 	- filter out any ftp-based operating system installs 
> (they can wait). 
> Only picking on this example because I did it this morning.
> 	What we have to be careful not to do is filter out 
> things that are
> higher priority than the priority voice conversation.  LSAs 
> (router link
> state announcements, necessary to keep the network converged and
> functioning) is one example.  
> 
> 
> -- 
> b
> 
> 
> _______________________________________________
> 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] RE:MLPPoIP - Preemption of low priority calls
From: Rex Buddenberg <budden@nps.navy.mil>
To: "Rosen, Brian" <Brian.Rosen@marconi.com>
Cc: Mpierce1@aol.com, jmpolk@cisco.com, steves@shentel.net,
        jgunn@ix.netcom.com, carlberg@g11.org.uk, ieprep@ietf.org
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Brian,

There have been a lot of wannabe TCP replacements over the years.  The
CS landscape is littered with them.  Why didn't any catch on?

Well, a few did.  The transport layer functionality ended up inside the
application.  NFS is a good example of this.  But leaving that aside,
why has TCP endured so well?

I suppose part of the reason is that Vint Cerf et al did so well.

But I think that one of the reasons that none of the wannabes never
caught on is that they only have tackled one of the problems and I think
there are four:
	- reliable multicast.  More on this below.
	- built-in security.  Authentication of SYNs, for example.
	- QoS sensitivity (more than just congestion)
	- early open.  

Reliable multicast is a conceptually sticky problem and there is more
than one correct answer.  I can articulate at least two:
	- in a reliable  file transfer type of operation, you handle a late
joiner by scrolling back to the beginning of the operation because that
late joiner needs the entire file.  This resembles TCP's reliability
definition, but in a multicast environment.
	- in a common operational picture type of operation, the session may
have been in existence for an indefinitely long period of time.  A late
joiner may need a little history, but certainly not all of it.  But once
he's joined, he needs transport layer reliability of some sort.
	The problem is complicated by the fact that the only scalable reliable
transport systems are subscriber-oriented; originator-origin systems all
fail on ACK-implosion.

Early open means including real data with the opening SYN and ACK
packets in the three-way handshake.  In TCP, these are overhead-only
packets.  If you have a low-interactivity hop somewhere in your network
(e.g. a satellite throw), then the propagation time for this
interactivity adds up quickly ... right at the front of a session.  
	XTP is one transport protocol that allows early-open piggy backing; I'd
assume there are more.  If you see somebody flogging a Performance
Enhancement Proxy gadget, this is usually what's under the hood.  

QoS sensitivity.  The problem is sticky because there are several
variables and attitudes ... you can't take the bellhead approach of
simply counting Erlangs any more.
	- a lost packet, esp in a radio network, may not be a sign of
congestion.  So the TCP reactions to a lost packet are now wide of the
target.
	- 'fairness' needs definition.  TCP window, backoff, and slowstart
behavior are all socialistic -- all pigs are equal.  One of the points
here is that we need to be able to make some pigs more equal than
others.  CBR is a poor descriptor, IMHO, because it has to do with
determinism, not congestion -- poor fit in nethead thinking.  If you
want determinism, say so.  Bounded delay delivery is a synonym.  But
determinism is _not_ what we need to make voice work and it's bought
quite expensively in terms of bandwidth abuse.
	- there's no free lunch and there seem to be several ways to spawn
lose-lose situations.  Bandwidth efficiency, latency, jitter,
interactivity and determinism are all in tension -- if you optimize
latency/interactivity, you will pay for it in losses in bandwidth
efficiency.  The default behavior at the network layer is to optimize
bandwidth efficiency and take whatever hits are required in the other
categories.  
	One of the current research projects at MIT is called XCP and is
designed to smarten up the qos impairment problem.  It requires
modifying the network layer protocols so that more information than
simply a lost packet is passed to the transport layer.  There are some
clever insights here, but I'm guessing XCP will be doomed to be a toy
unless it accounts for the other three bullets on the next-gen list.  



On Mon, 2003-12-01 at 11:06, Rosen, Brian wrote:
> This discussion relates to another thread happening on dccp.
> The basis of that thread is a discussion of whether it is possible
> to design a protocol that can be used for real time data that is
> "fair" with respect to TCP.  One of the opinions on this thread
> (mine among others) is that fairness can only be achieved if TCP 
> also changes to be aware of "constant bit rate"-like streams.  
> If there was a mechanism introduced into TCP that worked to allow 
> real time media and TCP to better co-exist, as well as protocols 
> that allowed media streams to better co-exist with TCP, then perhaps 
> there would be opportunities to address some of the concerns Rex raises.
> 
> Please also let me comment on:
> > 	- cut back a VTC to voice only.  The psychologists tell 
> > us that a
> > talking-head VTC has visual cues that we use for about 5-10 minutes;
> > after that we might as well turn the video off and just leave the
> > audio.  
> I'd like to see the citation on this research.  Our research shows that
> people in the same room with one another look intently on the faces of
> those talking, typically 90% of the time if engaged.  On a well designed
> "VTC", you can observe the same effect, even an hour or more into a call.  
> On a poorer quality system, participants tend to look around the room a 
> lot.  I wonder if the research you refer to is out of date.
> 
> And yes Scott, a router can't be expected to tell the difference between
> a well designed and a poorly designed video system. 
> 
> Brian
> 
> > -----Original Message-----
> > From: Rex Buddenberg [mailto:budden@nps.navy.mil]
> > Sent: Wednesday, November 26, 2003 4:47 PM
> > To: Mpierce1@aol.com
> > Cc: jmpolk@cisco.com; steves@shentel.net; jgunn@ix.netcom.com;
> > carlberg@g11.org.uk; ieprep@ietf.org
> > Subject: Re: [Ieprep] RE:MLPPoIP - Preemption of low priority calls
> > 
> > 
> > On Wed, 2003-11-26 at 12:17, Mpierce1@aol.com wrote:
> > ...
> > > 
> > > The current GSCR was written for circuit mode. Unless we get it
> > > changed, it will apply to packet mode (even when it doesn't always
> > > make sense). This is one of the areas we are trying to change, so it
> > > correctly takes into account the advantages of using packet. 
> > ...
> > 
> > 
> > This whole conversation appears to be predicated on some 
> > tacit and false
> > assumptions.
> > 
> > The principal assumption seems to be that the entire capacity of a hop
> > is taken up with VOICE conversations so that the solution to
> > oversubscription is to shed some of those VOICE conversations.  
> > 
> > If you come to the conversation with a bellhead mentality, then we can
> > probably read up on Erlangs and reach much the same 
> > conclusion that the
> > circuit switch world reached years ago.
> > 
> > 
> > If you have a priority voice conversation and experience 
> > congestion, it
> > would seem to me that we would want to bump something other than other
> > voice conversations:
> > 	- anything file transfer (incl e-mail enclosures) with 
> > .ppt in the file
> > label.  The content:bit-bloat ratio there is terrible to the 
> > extent that
> > I'm calling powerpoint a virus these days.
> > 	- cut back a VTC to voice only.  The psychologists tell 
> > us that a
> > talking-head VTC has visual cues that we use for about 5-10 minutes;
> > after that we might as well turn the video off and just leave the
> > audio.  The problem here is that we can't know by looking at port
> > numbers whether someone has turned the camera around and is 
> > sharing view
> > of a crime scene
> > 	- filter out any ftp-based operating system installs 
> > (they can wait). 
> > Only picking on this example because I did it this morning.
> > 	What we have to be careful not to do is filter out 
> > things that are
> > higher priority than the priority voice conversation.  LSAs 
> > (router link
> > state announcements, necessary to keep the network converged and
> > functioning) is one example.  
> > 
> > 
> > -- 
> > b
> > 
> > 
> > _______________________________________________
> > 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
-- 
b


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



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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Rex Buddenberg'" <budden@nps.navy.mil>
Cc: Mpierce1@aol.com, jmpolk@cisco.com, steves@shentel.net,
        jgunn@ix.netcom.com, carlberg@g11.org.uk, ieprep@ietf.org
Subject: RE: [Ieprep] RE:MLPPoIP - Preemption of low priority calls
Date: Mon, 1 Dec 2003 15:16:59 -0500 
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Rex

I agree with all you have written, and yet I wonder if the problem
we are having getting VoIP and TCP to co-exist nicely is going to
be solved if we try to also solve the other problems you raise.
It would be nice, I think, to do so, but is it practical?

It does seem that we could tinker with TCP to allow it to be
less greedy than it is now, just as we can tinker with UDP to add
enough congestion control to prevent congestion collapse and to
appropriately trigger better TCP behavior.  Some of the proposals on
the DCCP list are, uh, interesting.  Like the one that suggests
that we use up to twice the bandwidth we need so that TCP will back
off enough to maintain what we need.  That would actually work, but
it may not be the best way to solve the problem.

On this list, the issue is do we actually have some requirements
in IEPREP that impinge on TCP that are similar to, but not exactly
the same as, the VoIP (or other real-time media) problem?  
Should we codify such requirements so that IEPREP could be added to 
the (small) chorus of voices that are talking about heresy -- making 
TCP not so greedy when no one else is pigging out.

I treat TCP replacements as "run away, run away fast" for much of the
reasons you have elucidated.  However, we have tinkered with TCP in
several ways of late.  Not all have (yet) been widely implemented,
but there is some hope.  The problem at the moment is that the UDP
toting media stream folks don't have any incentive to add congestion
control when the big bad TCP gobbles up all their bandwidth if they 
do or they don't, and the TCP crowd thinks if everyone else was just
TCP friendly (they even have an acronym!) life would be great, for TCP.

Brian

> -----Original Message-----
> From: Rex Buddenberg [mailto:budden@nps.navy.mil]
> Sent: Monday, December 01, 2003 2:45 PM
> To: Rosen, Brian
> Cc: Mpierce1@aol.com; jmpolk@cisco.com; steves@shentel.net;
> jgunn@ix.netcom.com; carlberg@g11.org.uk; ieprep@ietf.org
> Subject: RE: [Ieprep] RE:MLPPoIP - Preemption of low priority calls
> 
> 
> Brian,
> 
> There have been a lot of wannabe TCP replacements over the years.  The
> CS landscape is littered with them.  Why didn't any catch on?
> 
> Well, a few did.  The transport layer functionality ended up 
> inside the
> application.  NFS is a good example of this.  But leaving that aside,
> why has TCP endured so well?
> 
> I suppose part of the reason is that Vint Cerf et al did so well.
> 
> But I think that one of the reasons that none of the wannabes never
> caught on is that they only have tackled one of the problems 
> and I think
> there are four:
> 	- reliable multicast.  More on this below.
> 	- built-in security.  Authentication of SYNs, for example.
> 	- QoS sensitivity (more than just congestion)
> 	- early open.  
> 
> Reliable multicast is a conceptually sticky problem and there is more
> than one correct answer.  I can articulate at least two:
> 	- in a reliable  file transfer type of operation, you 
> handle a late
> joiner by scrolling back to the beginning of the operation 
> because that
> late joiner needs the entire file.  This resembles TCP's reliability
> definition, but in a multicast environment.
> 	- in a common operational picture type of operation, 
> the session may
> have been in existence for an indefinitely long period of 
> time.  A late
> joiner may need a little history, but certainly not all of 
> it.  But once
> he's joined, he needs transport layer reliability of some sort.
> 	The problem is complicated by the fact that the only 
> scalable reliable
> transport systems are subscriber-oriented; originator-origin 
> systems all
> fail on ACK-implosion.
> 
> Early open means including real data with the opening SYN and ACK
> packets in the three-way handshake.  In TCP, these are overhead-only
> packets.  If you have a low-interactivity hop somewhere in 
> your network
> (e.g. a satellite throw), then the propagation time for this
> interactivity adds up quickly ... right at the front of a session.  
> 	XTP is one transport protocol that allows early-open 
> piggy backing; I'd
> assume there are more.  If you see somebody flogging a Performance
> Enhancement Proxy gadget, this is usually what's under the hood.  
> 
> QoS sensitivity.  The problem is sticky because there are several
> variables and attitudes ... you can't take the bellhead approach of
> simply counting Erlangs any more.
> 	- a lost packet, esp in a radio network, may not be a sign of
> congestion.  So the TCP reactions to a lost packet are now wide of the
> target.
> 	- 'fairness' needs definition.  TCP window, backoff, 
> and slowstart
> behavior are all socialistic -- all pigs are equal.  One of the points
> here is that we need to be able to make some pigs more equal than
> others.  CBR is a poor descriptor, IMHO, because it has to do with
> determinism, not congestion -- poor fit in nethead thinking.  If you
> want determinism, say so.  Bounded delay delivery is a synonym.  But
> determinism is _not_ what we need to make voice work and it's bought
> quite expensively in terms of bandwidth abuse.
> 	- there's no free lunch and there seem to be several 
> ways to spawn
> lose-lose situations.  Bandwidth efficiency, latency, jitter,
> interactivity and determinism are all in tension -- if you optimize
> latency/interactivity, you will pay for it in losses in bandwidth
> efficiency.  The default behavior at the network layer is to optimize
> bandwidth efficiency and take whatever hits are required in the other
> categories.  
> 	One of the current research projects at MIT is called XCP and is
> designed to smarten up the qos impairment problem.  It requires
> modifying the network layer protocols so that more information than
> simply a lost packet is passed to the transport layer.  There are some
> clever insights here, but I'm guessing XCP will be doomed to be a toy
> unless it accounts for the other three bullets on the next-gen list.  
> 
> 
> 
> On Mon, 2003-12-01 at 11:06, Rosen, Brian wrote:
> > This discussion relates to another thread happening on dccp.
> > The basis of that thread is a discussion of whether it is possible
> > to design a protocol that can be used for real time data that is
> > "fair" with respect to TCP.  One of the opinions on this thread
> > (mine among others) is that fairness can only be achieved if TCP 
> > also changes to be aware of "constant bit rate"-like streams.  
> > If there was a mechanism introduced into TCP that worked to allow 
> > real time media and TCP to better co-exist, as well as protocols 
> > that allowed media streams to better co-exist with TCP, 
> then perhaps 
> > there would be opportunities to address some of the 
> concerns Rex raises.
> > 
> > Please also let me comment on:
> > > 	- cut back a VTC to voice only.  The psychologists tell 
> > > us that a
> > > talking-head VTC has visual cues that we use for about 
> 5-10 minutes;
> > > after that we might as well turn the video off and just leave the
> > > audio.  
> > I'd like to see the citation on this research.  Our 
> research shows that
> > people in the same room with one another look intently on 
> the faces of
> > those talking, typically 90% of the time if engaged.  On a 
> well designed
> > "VTC", you can observe the same effect, even an hour or 
> more into a call.  
> > On a poorer quality system, participants tend to look 
> around the room a 
> > lot.  I wonder if the research you refer to is out of date.
> > 
> > And yes Scott, a router can't be expected to tell the 
> difference between
> > a well designed and a poorly designed video system. 
> > 
> > Brian
> > 
> > > -----Original Message-----
> > > From: Rex Buddenberg [mailto:budden@nps.navy.mil]
> > > Sent: Wednesday, November 26, 2003 4:47 PM
> > > To: Mpierce1@aol.com
> > > Cc: jmpolk@cisco.com; steves@shentel.net; jgunn@ix.netcom.com;
> > > carlberg@g11.org.uk; ieprep@ietf.org
> > > Subject: Re: [Ieprep] RE:MLPPoIP - Preemption of low 
> priority calls
> > > 
> > > 
> > > On Wed, 2003-11-26 at 12:17, Mpierce1@aol.com wrote:
> > > ...
> > > > 
> > > > The current GSCR was written for circuit mode. Unless we get it
> > > > changed, it will apply to packet mode (even when it 
> doesn't always
> > > > make sense). This is one of the areas we are trying to 
> change, so it
> > > > correctly takes into account the advantages of using packet. 
> > > ...
> > > 
> > > 
> > > This whole conversation appears to be predicated on some 
> > > tacit and false
> > > assumptions.
> > > 
> > > The principal assumption seems to be that the entire 
> capacity of a hop
> > > is taken up with VOICE conversations so that the solution to
> > > oversubscription is to shed some of those VOICE conversations.  
> > > 
> > > If you come to the conversation with a bellhead 
> mentality, then we can
> > > probably read up on Erlangs and reach much the same 
> > > conclusion that the
> > > circuit switch world reached years ago.
> > > 
> > > 
> > > If you have a priority voice conversation and experience 
> > > congestion, it
> > > would seem to me that we would want to bump something 
> other than other
> > > voice conversations:
> > > 	- anything file transfer (incl e-mail enclosures) with 
> > > .ppt in the file
> > > label.  The content:bit-bloat ratio there is terrible to the 
> > > extent that
> > > I'm calling powerpoint a virus these days.
> > > 	- cut back a VTC to voice only.  The psychologists tell 
> > > us that a
> > > talking-head VTC has visual cues that we use for about 
> 5-10 minutes;
> > > after that we might as well turn the video off and just leave the
> > > audio.  The problem here is that we can't know by looking at port
> > > numbers whether someone has turned the camera around and is 
> > > sharing view
> > > of a crime scene
> > > 	- filter out any ftp-based operating system installs 
> > > (they can wait). 
> > > Only picking on this example because I did it this morning.
> > > 	What we have to be careful not to do is filter out 
> > > things that are
> > > higher priority than the priority voice conversation.  LSAs 
> > > (router link
> > > state announcements, necessary to keep the network converged and
> > > functioning) is one example.  
> > > 
> > > 
> > > -- 
> > > b
> > > 
> > > 
> > > _______________________________________________
> > > 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
> -- 
> b
> 
> 
> _______________________________________________
> Ieprep mailing list
> Ieprep@ietf.org
> https://www1.ietf.org/mailman/listinfo/ieprep
> 

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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		: IP Telephony Requirements for Emergency Telecommunication Service
	Author(s)	: K. Carlberg, R. Atkinson
	Filename	: draft-ietf-ieprep-ets-telephony-07.txt
	Pages		: 6
	Date		: 2003-12-2
	
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 [3].
Solutions to these requirements are not presented in this document.

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FYI

-----Original Message-----
From: owner-ietf-announce@ietf.org
[mailto:owner-ietf-announce@ietf.org]On Behalf Of
Internet-Drafts@ietf.org
Sent: Tuesday, December 02, 2003 3:35 PM
Subject: I-D ACTION:draft-baker-tsvwg-mlpp-that-works-00.txt


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


	Title		: Implementing MLPP in the Internet Protocol Suite
	Author(s)	: F. Baker
	Filename	: draft-baker-tsvwg-mlpp-that-works-00.txt
	Pages		: 23
	Date		: 2003-12-2
	
The Defense Information Systems Agency of the United States
Department of Defense, with is contractors, has proposed a service
architecture for military (NATO and related agencies) telephone
systems. This is called the Assured Service, and is defined in two
documents: 'Architecture for Assured Service Capabilities in Voice
over IP' and 'Requirements for Assured Service Capabilities in Voice
over IP'. Responding to these are three documents: 'Reason Header   Field for the Session Initiation Protocol', 'Extending the Session
Initiation Protocol Reason Header to account for  Preemption Events',
'Communications Resource Priority for the Session Initiation
Protocol'.

   What remains to this specification is to provide a Call Admission
Control procedure and a Per Hop Behavior for the data which meet the
   needs of this architecture.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-baker-tsvwg-mlpp-that-works-00.txt

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-----Original Message-----
From: owner-ietf-announce@ietf.org
[mailto:owner-ietf-announce@ietf.org]On Behalf Of
Internet-Drafts@ietf.org
Sent: Tuesday, December 02, 2003 3:35 PM
Subject: I-D ACTION:draft-baker-tsvwg-mlef-concerns-00.txt


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


	Title		: MLEF Considered Harmful
	Author(s)	: F. Baker
	Filename	: draft-baker-tsvwg-mlef-concerns-00.txt
	Pages		: 19
	Date		: 2003-12-2
	
The Defense Information Systems Agency of the United States
Department of Defense, with is contractors, has proposed a service
architecture for military (NATO and related agencies) telephone
systems. This is called the Assured Service, and is defined in two
documents: 'Architecture for Assured Service Capabilities in Voice
over IP' and 'Requirements for Assured Service Capabilities in Voice
over IP'. Responding to these are four documents: 'Reason Header
Field for the Session Initiation Protocol', 'Extending the Session
Initiation Protocol Reason Header to account for  Preemption Events',
'Communications Resource Priority for the Session Initiation
Protocol', and the 'Multi-Level Expedited Forwarding Per Hop
Behavior' (MLEF PHB). MLEF, as currently defined, has serious
problems, which this draft seeks to discuss.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-baker-tsvwg-mlef-concerns-00.txt

To remove yourself from the IETF Announcement list, send a message to 
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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 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-07.txt
	Pages		: 26
	Date		: 2003-12-3
	
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.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-ieprep-framework-07.txt

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From: The IESG <iesg-secretary@ietf.org>
To: IETF-Announce:;
Cc: Internet Architecture Board <iab@iab.org>,
        RFC Editor <rfc-editor@rfc-editor.org>, <ieprep@ietf.org>
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Date: Thu, 11 Dec 2003 20:42:04 -0500
Subject: [Ieprep] Document Action: 'IP Telephony Requirements for Emergency
 Telecommunication Service' to Informational RFC
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The IESG has approved the following document:

- 'IP Telephony Requirements for Emergency Telecommunication Service '
   <draft-ietf-ieprep-ets-telephony-07.txt> as an Informational RFC

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

The IESG contact persons are Jon Peterson and Allison Mankin.

Technical Summary

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 previously developed in the IEPREP WG. Solutions to these requirements are not presented in this document.
 
Working Group Summary
 
The IEPREP Working Group supports the document.
 
Protocol Quality
 
This document was reviewed for the IESG by Jon Peterson.


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From: "King, Kimberly  S." <KIMBERLY.S.KING@saic.com>
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Subject: [Ieprep] TCP PEPs and Priority-based Mechanisms
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This Information RFC may be of interest.  For example, PEPs can be useful 
for moving TCP traffic over satellite links.

http://www.ietf.org/rfc/rfc3135.txt

3.5 Priority-based Multiplexing

   Implementing priority-based multiplexing of data over a slow and
   expensive link may significantly improve the performance and
   usability of the link for selected applications or connections.

   A user behind a slow link would experience the link more feasible to
   use in case of simultaneous data transfers, if urgent data transfers
   (e.g., interactive connections) could have shorter response time
   (better performance) than less urgent background transfers.  If the
   interactive connections transmit enough data to keep the slow link
   fully utilized, it might be necessary to fully suspend the background
   transfers for awhile to ensure timely delivery for the interactive
   connections.

   In flight TCP segments of an end-to-end TCP connection (with low
   priority) cannot be delayed for a long time.  Otherwise, the TCP
   timer at the sending end would expire, resulting in suboptimal
   performance.  However, this kind of operation can be controlled in
   conjunction with a split connection TCP PEP by assigning different
   priorities for different connections (or applications).  A split
   connection PEP implementation allows the PEP in an intermediate node
   to delay the data delivery of a lower-priority TCP flow for an
   unlimited period of time by simply rescheduling the order in which it
   forwards data of different flows to the destination host behind the
   slow link.  This does not have a negative impact on the delayed TCP
   flow as normal TCP flow control takes care of suspending the flow
   between the TCP sender and the PEP, when the PEP is not forwarding
   data for the flow, and resumes it once the PEP decides to continue
   forwarding data for the flow.  This can further be assisted, if the
   protocol stacks on both sides of the slow link implement priority
   based scheduling of connections.

   With such a PEP implementation, along with user-controlled
   priorities, the user can assign higher priority for selected
   interactive connection(s) and have much shorter response time for the
   selected connection(s), even if there are simultaneous low priority
   bulk data transfers which in regular end-to-end operation would
   otherwise eat the available bandwidth of the slow link almost
   completely.  These low priority bulk data transfers would then
   proceed nicely during the idle periods of interactive connections,
   allowing the user to keep the slow and expensive link (e.g., wireless
   WAN) fully utilized.

   Other priority-based mechanisms may be applied on shared wireless
   links with more than two terminals.  With shared wireless mediums
   becoming a weak link in Internet QoS architectures, many may turn to
   PEPs to provide extra priority levels across a shared wireless medium
   [SHEL00].  These PEPs are distributed on all nodes of the shared
   wireless medium.  For example, in an 802.11 WLAN this PEP is
   implemented in the access point (base station) and each mobile host.
   One PEP then uses distributed queuing techniques to coordinate
   traffic classes of all nodes.  This is also sometimes called subnet
   bandwidth management.  See [BBKT97] for an example of queuing
   techniques which can be used to achieve this.  This technique can be
   implemented either above or below the IP layer.  Priority treatment
   can typically be specified either by the user or by marking the
   (IPv4) ToS or (IPv6) Traffic Class IP header field.

_______________________________________________
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https://www1.ietf.org/mailman/listinfo/ieprep



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Subject: Re: [Ieprep] TCP PEPs and Priority-based Mechanisms
From: Rex Buddenberg <budden@nps.navy.mil>
To: "King, Kimberly  S." <KIMBERLY.S.KING@saic.com>
Cc: "Ieprep (E-mail)" <ieprep@ietf.org>
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Make sure you read section 6 on page 33.

The use of Performance Enhancing Proxies introduces several issues
   which impact security.  First, (as described in detail in Section
   4.1.1,) using PEPs and using IPsec is generally mutually exclusive.



On Mon, 2003-12-22 at 08:10, King, Kimberly S. wrote:
> This Information RFC may be of interest.  For example, PEPs can be useful 
> for moving TCP traffic over satellite links.
> 
> http://www.ietf.org/rfc/rfc3135.txt
> 
> 3.5 Priority-based Multiplexing
> 
>    Implementing priority-based multiplexing of data over a slow and
>    expensive link may significantly improve the performance and
>    usability of the link for selected applications or connections.
> 
>    A user behind a slow link would experience the link more feasible to
>    use in case of simultaneous data transfers, if urgent data transfers
>    (e.g., interactive connections) could have shorter response time
>    (better performance) than less urgent background transfers.  If the
>    interactive connections transmit enough data to keep the slow link
>    fully utilized, it might be necessary to fully suspend the background
>    transfers for awhile to ensure timely delivery for the interactive
>    connections.
> 
>    In flight TCP segments of an end-to-end TCP connection (with low
>    priority) cannot be delayed for a long time.  Otherwise, the TCP
>    timer at the sending end would expire, resulting in suboptimal
>    performance.  However, this kind of operation can be controlled in
>    conjunction with a split connection TCP PEP by assigning different
>    priorities for different connections (or applications).  A split
>    connection PEP implementation allows the PEP in an intermediate node
>    to delay the data delivery of a lower-priority TCP flow for an
>    unlimited period of time by simply rescheduling the order in which it
>    forwards data of different flows to the destination host behind the
>    slow link.  This does not have a negative impact on the delayed TCP
>    flow as normal TCP flow control takes care of suspending the flow
>    between the TCP sender and the PEP, when the PEP is not forwarding
>    data for the flow, and resumes it once the PEP decides to continue
>    forwarding data for the flow.  This can further be assisted, if the
>    protocol stacks on both sides of the slow link implement priority
>    based scheduling of connections.
> 
>    With such a PEP implementation, along with user-controlled
>    priorities, the user can assign higher priority for selected
>    interactive connection(s) and have much shorter response time for the
>    selected connection(s), even if there are simultaneous low priority
>    bulk data transfers which in regular end-to-end operation would
>    otherwise eat the available bandwidth of the slow link almost
>    completely.  These low priority bulk data transfers would then
>    proceed nicely during the idle periods of interactive connections,
>    allowing the user to keep the slow and expensive link (e.g., wireless
>    WAN) fully utilized.
> 
>    Other priority-based mechanisms may be applied on shared wireless
>    links with more than two terminals.  With shared wireless mediums
>    becoming a weak link in Internet QoS architectures, many may turn to
>    PEPs to provide extra priority levels across a shared wireless medium
>    [SHEL00].  These PEPs are distributed on all nodes of the shared
>    wireless medium.  For example, in an 802.11 WLAN this PEP is
>    implemented in the access point (base station) and each mobile host.
>    One PEP then uses distributed queuing techniques to coordinate
>    traffic classes of all nodes.  This is also sometimes called subnet
>    bandwidth management.  See [BBKT97] for an example of queuing
>    techniques which can be used to achieve this.  This technique can be
>    implemented either above or below the IP layer.  Priority treatment
>    can typically be specified either by the user or by marking the
>    (IPv4) ToS or (IPv6) Traffic Class IP header field.
> 
> _______________________________________________
> Ieprep mailing list
> Ieprep@ietf.org
> https://www1.ietf.org/mailman/listinfo/ieprep
-- 
b


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To: "'Rex Buddenberg'" <budden@nps.navy.mil>,
        "King, Kimberly  S." <KIMBERLY.S.KING@saic.com>
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Subject: RE: [Ieprep] TCP PEPs and Priority-based Mechanisms
Date: Mon, 22 Dec 2003 14:16:48 -0500
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> The use of Performance Enhancing Proxies introduces several issues
>    which impact security.=20

Yes, you are right to point that out but unfortunately that's just the =
way
it is.  Even if we forget about PEPs for the moment. =20

Use IPsec but mark the packets using DSCPs kept in the clear.  Consider
traffic analysis...sit back and watch the codepoints (lots of high =
priority
MLEF?)

"
* Security UP =3D>=AD Bandwidth Utilization DOWN
* Security UP =3D>=AD Performance DOWN
* Tunnels Tunnels Tunnels and more Tunnels
* Performance DOWN =3D> Security DOWN
=3D>  User turns OFF Security to make system usable!
* Thus, we need more bandwidth to ensure security.
"
http://spacecom.grc.nasa.gov/icnsconf/docs/2002/10/Session_E1-5_Ivancic.=
pdf

I think one makes the choices that best fit because there is no free =
lunch.
:-(


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Subject: RE: [Ieprep] TCP PEPs and Priority-based Mechanisms
From: Rex Buddenberg <budden@nps.navy.mil>
To: "King, Kimberly  S." <KIMBERLY.S.KING@saic.com>
Cc: "Ieprep (E-mail)" <ieprep@ietf.org>
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Kimberly,

Actually you hit the one compelling reason, IMHO, for shifting to IPv6.=20
Note that I'm here at the network layer, not transport.  IPv6 supports
header extensions -- the extension+payload get encrypted but the base
header (which includes DSCP) does not.  So any network level QoS you
choose to use is unaffected by IPSec.

Back at the transport layer, I'm just not terribly turned on by PEPs.=20
The RFC is useful because it surveys a fairly wide set of bandaids that
fall under the general PEP category.   But these remain bandaids until
we grapple with the nextgen transport protocol issue.


On Mon, 2003-12-22 at 11:16, King, Kimberly S. wrote:
> > The use of Performance Enhancing Proxies introduces several issues
> >    which impact security.=20
>=20
> Yes, you are right to point that out but unfortunately that's just the =
way
> it is.  Even if we forget about PEPs for the moment. =20
>=20
> Use IPsec but mark the packets using DSCPs kept in the clear.  Consider
> traffic analysis...sit back and watch the codepoints (lots of high prio=
rity
> MLEF?)
>=20
> "
> * Security UP =3D>=C2=AD Bandwidth Utilization DOWN
> * Security UP =3D>=C2=AD Performance DOWN
> * Tunnels Tunnels Tunnels and more Tunnels
> * Performance DOWN =3D> Security DOWN
> =3D>  User turns OFF Security to make system usable!
> * Thus, we need more bandwidth to ensure security.
> "
> http://spacecom.grc.nasa.gov/icnsconf/docs/2002/10/Session_E1-5_Ivancic=
.pdf
>=20
> I think one makes the choices that best fit because there is no free lu=
nch.
> :-(
--=20
b


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Subject: RE: [Ieprep] TCP PEPs and Priority-based Mechanisms
Date: Mon, 22 Dec 2003 14:55:50 -0600
Message-ID: <5EF7D95E17BDAD4A968C812E5ABC390B011081F0@KC-MAIL4.kc.umkc.edu>
Thread-Topic: [Ieprep] TCP PEPs and Priority-based Mechanisms
Thread-Index: AcPIxRQLgO1Kdof8TtqG1sNvNNuyLAABY/Ji
From: "Ayyasamy, Senthilkumar  \(UMKC-Student\)" <saq66@umkc.edu>
To: "Rex Buddenberg" <budden@nps.navy.mil>,
        "King, Kimberly  S." <KIMBERLY.S.KING@saic.com>
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> Actually you hit the one compelling reason, IMHO, for shifting to =
IPv6.
> Note that I'm here at the network layer, not transport.  IPv6 supports
> header extensions -- the extension+payload get encrypted but the base
> header (which includes DSCP) does not.  So any network level QoS you
> choose to use is unaffected by IPSec.
=20
This will lead to denial of QoS attacks. One can easily change the=20
subscription by modifying the headers. But, lets first see some=20
flow-label enabled packets across the Internet to talk about the=20
operational security implications. btw, one can still encrypt=20
flow-labels using tunnel mode. please read the recently approved
proposed standard ( IPv6 flow label spec)

> Back at the transport layer, I'm just not terribly turned on by PEPs.
> The RFC is useful because it surveys a fairly wide set of bandaids =
that
> fall under the general PEP category.   But these remain bandaids until
> we grapple with the nextgen transport protocol issue.
=20
nextgen transport protocol? DCCP is not a next generation protocol; it=20
just provides option for various cc profiles, mibility and multihoming
support and security features. As for tranport, incremental approach by=20
TCP extenstions works better (e.g., widespread SACK deployment.) The one =

you mention, XCP does major changes; its a interesting research work but =

support for high speed networks is not a pressing problem. In short, we=20
will never see any nextgen transport protocol.
=20
btw, protocol boosters like PEP are not paper model; they have seen a =
lot=20
of deployment. So, we should know to live with them. We have came a long
way from the pilc work.=20

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testing 1 2 3

cheers,
James

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


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Date: Mon, 22 Dec 2003 17:08:02 -0600
To: Rex Buddenberg <budden@nps.navy.mil>
From: "James M. Polk" <jmpolk@cisco.com>
Subject: RE: [Ieprep] TCP PEPs and Priority-based Mechanisms
Cc: "Ieprep (E-mail)" <ieprep@ietf.org>
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At 11:50 AM 12/22/2003 -0800, Rex Buddenberg wrote:
>Kimberly,
>
>Actually you hit the one compelling reason, IMHO, for shifting to IPv6.
>Note that I'm here at the network layer, not transport.  IPv6 supports
>header extensions -- the extension+payload get encrypted but the base
>header (which includes DSCP) does not.  So any network level QoS you
>choose to use is unaffected by IPSec.

How does IPv6 behave differently than v4 with respect to IPsec?

What do you do to limit or prevent remarking DSCPs to mean something other=
=20
than what the sender (or domain at the edge) legitimately mean?

In other words, where is the integrity protection for DSCP marking? Where=20
is the authorization mechanism for DSCP marking?

How do you prevent an eavesdropper from looking for the pattern of packets=
=20
(perhaps with the same length) occurring every 20ms or so to see if a=20
particular stream is a call or not (perhaps to do something to that stream=
=20
- or just the IPv4 or v6 header to muck with it (like change the DSCP=
 value))?


>Back at the transport layer, I'm just not terribly turned on by PEPs.
>The RFC is useful because it surveys a fairly wide set of bandaids that
>fall under the general PEP category.   But these remain bandaids until
>we grapple with the nextgen transport protocol issue.
>
>
>On Mon, 2003-12-22 at 11:16, King, Kimberly S. wrote:
> > > The use of Performance Enhancing Proxies introduces several issues
> > >    which impact security.
> >
> > Yes, you are right to point that out but unfortunately that's just the=
 way
> > it is.  Even if we forget about PEPs for the moment.
> >
> > Use IPsec but mark the packets using DSCPs kept in the clear.  Consider
> > traffic analysis...sit back and watch the codepoints (lots of high=
 priority
> > MLEF?)
> >
> > "
> > * Security UP =3D>=C2=AD Bandwidth Utilization DOWN
> > * Security UP =3D>=C2=AD Performance DOWN
> > * Tunnels Tunnels Tunnels and more Tunnels
> > * Performance DOWN =3D> Security DOWN
> > =3D>  User turns OFF Security to make system usable!
> > * Thus, we need more bandwidth to ensure security.
> > "
> >=
 http://spacecom.grc.nasa.gov/icnsconf/docs/2002/10/Session_E1-5_Ivancic.pdf
> >
> > I think one makes the choices that best fit because there is no free=
 lunch.
> > :-(
>--
>b
>
>
>_______________________________________________
>Ieprep mailing list
>Ieprep@ietf.org
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cheers,
James

                                *******************
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From: "Eagan, Christopher" <christopher.eagan@lmco.com>
Subject: RE: [Ieprep] TCP PEPs and Priority-based Mechanisms
To: "James M. Polk" <jmpolk@cisco.com>, Rex Buddenberg <budden@nps.navy.mil>
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> What do you do to limit or prevent remarking DSCPs to mean 
> something other 
> than what the sender (or domain at the edge) legitimately mean?

1.  In some scenarios can you assume all nodes are trusted to behave well (possibly by virtue of them being able to encrypt/decrypt at a lower network layer - "they can speak on this encrypted link layer therefore I trust them at the higher layers")?  Obviously this does not satisfy the problems for security purely in the internet layer but may be practical.

2.  Has anyone considered dynamic and pseudo-random codepoints?  Ingress nodes sharing time-based pseudo-randomly sequenced dscp->traffic class mappings - starting values updated periodically generated from a real random sequence?  This will promote the precedence of some packets that should not be promoted but all in all could work?  Anybody looked at this?  Too organizationally infeasible for IANA perhaps?

> In other words, where is the integrity protection for DSCP 
> marking? Where 
> is the authorization mechanism for DSCP marking?
> How do you prevent an eavesdropper from looking for the 
> pattern of packets 
> (perhaps with the same length) occurring every 20ms or so to see if a 

The issues of unencrypted headers being altered and traffic analysis are not unique to dscp's.  Again these can both be solved by requiring lower layer bulk encryption and requiring lower layers to send random junk when there is no legitimate traffic (so the observer sees a random stream of bits at all times independent of the actual traffic).  Not satisfying the security requirements in the internet layer but may be most practical solution.

Chris

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