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From: Ron Bonica <rbonica@juniper.net>
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Subject: [tcpm] draft-bonica-tcp-auth-04
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Folks,

Please review the draft at the following URL:

   http://www.ietf.org/internet-drafts/draft-bonica-tcp-auth-04.txt

An abstract follows:

   "This memo describes a TCP extension that enhances security for BGP,
   LDP and other TCP-based protocols.  It is intended for applications
   where secure administrative access to both the end-points of the TCP
   connection is normally available.  TCP peers can use this extension
   to authenticate messages passed between one another.

   The strategy described herein improves upon current practice, which
   is described in RFC 2385.  Using this new strategy, TCP peers can
   update authentication keys during the lifetime of a TCP connection.
   TCP peers can also use stronger authentication algorithms to
   authenticate routing messages."

                                   Ron

	



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From tcpm-bounces@ietf.org Mon Feb 06 19:55:45 2006
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Date: Mon, 06 Feb 2006 19:54:51 -0500
From: "Armando L. Caro, Jr." <acaro@bbn.com>
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To: Janardhan Iyengar <iyengar@mail.eecis.udel.edu>
Subject: Re: [tcpm] draft on congestion window limiting
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	<Pine.GSO.4.62.0601301630570.589@stimpy.eecis.udel.edu>
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Jana,

I finally got around to reading this draft. I don't have much to add to
the discussion. I agree that experimenting should be done. I wonder how
much micro-bursts are a concern given that macro-bursts exist. Are
micro-bursts just a little bit of noise in the bigger scheme of things,
or do they really add up when you aggregate many TCP flows?

I also have some comments below...

Janardhan Iyengar wrote:

> As far as the buggy receiver is concerned, I'm not sure that we should
> aim to demonstrate trade-offs in such cases. But, IIUC, your core point
> is that under the same conditions, a sender with burst-mitigation might
> be at a disadvantage with respect to a sender without burst-mitigation.
> There is also the possibility that a sender without burst-mitigation
> loses packets in the network due to the burst when a sender with
> burst-mitigation does not. So, I agree that there is a tradeoff. And
> that we should write it out clearly.

Well, this tradeoff is fairly obvious I think. You could include for
completeness I suppose. I'm not too concerned about this topic, because
we always face this tradeoff when dealing with correct congestion
control. Correct congestion control almost always trades off a sender's
throughput with the network's health. So, if sender's wanted to be
greedy, they COULD avoid congestion control altogether. However, we
don't see this as a default behavior in major operating systems.

Nit picky comments...

First sentence in Section 2: your missing the word "sender" after "TCP".

Second paragraph in Section 3: I would end add another sentence to the
end of the paragraph: "Also, CWV is unnecessary if CWL is used."

Second paragraph in the first bullet of Section 3: I find this paragraph
confusing. The sentence "An additional drawback..." seems out of place
here. What's the other drawback? The beginning of the paragraph goes
back and forth of whether two controls are good or not, so this sentence
threw me off.

Again, I think that experimenting should be done to determine whether
micro-bursts are an issue or not. Unfortunately, I don't have
suggestions of where to start. I'll need to think more on that topic...

-- 
Armando


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From tcpm-bounces@ietf.org Tue Feb 07 15:36:33 2006
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To: Janardhan Iyengar <iyengar@mail.eecis.udel.edu>
Subject: Re: [tcpm] draft on congestion window limiting
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Janardhan Iyengar wrote:
> Pasi,
> 
> Thank you for your comments, and for spending your time on this document.
> 
>> I believe experimenting with burst mitigation algorithms in real
>> networks could be possible for many of us. At least Linux has
>> implemented a similar scheme for years, though I don't know what's the
>> status of TCP in the latest kernel versions.
> 
> 
> Thank you for this bit of information. The scheme mentioned in the paper 
> looks like the "Use it or Lose it" scheme mentioned in the draft. I did 
> probe the Linux community recently about whether a micro-burst 
> mitigation scheme was deployed in Linux. The response I got was that 
> there wasn't any such scheme currently deployed, but patches were 
> available. I am not sure at this time if there is or isn't a scheme in 
> Linux (I haven't looked at the code myself yet). I would be happy if 
> someone could confirm for me which one is currently the case.

Linux does implement a somewhat similar scheme, with the primary 
differences being that it only applies the burst limit upon exit from a 
reorder/recovery state rather than on every ack, and it does not set ssthresh 
when reducing cwnd.  This actually makes its behavior substantially 
different.

One issue I see is if the delayed ack factor is greater than BLimit/MSS 
this breaks.  Or for that matter, if there are persistent stretch-acks, say, 
from a cpu-bound receiver or reordering on the reverse path.  Actually, I 
think persistent reordering on the forward path also causes problems.

I'm wondering if setting BLimit to the initial window may be too conservative.  
If I have a 1000 segment window, most likely a burst of 5 segments isn't 
going to hurt too much.

As for (dis)inscentives, some of this competing bursting/non-bursting senders 
issue is described in "Understanding the Performance of TCP Pacing" by 
Aggarwall, et. al. 
(http://www-cse.ucsd.edu/~savage/papers/Infocom2000pacing.pdf).  Though this 
isn't pacing as they describe it, I think some of the issues are very 
similar.

  -John

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Subject: Re: [tcpm] draft on congestion window limiting
From: Pasi Sarolahti <pasi.sarolahti@iki.fi>
To: "Armando L. Caro, Jr." <acaro@bbn.com>
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On Mon, 2006-02-06 at 19:54 -0500, Armando L. Caro, Jr. wrote:
> > As far as the buggy receiver is concerned, I'm not sure that we should
> > aim to demonstrate trade-offs in such cases. But, IIUC, your core point
> > is that under the same conditions, a sender with burst-mitigation might
> > be at a disadvantage with respect to a sender without burst-mitigation.
> > There is also the possibility that a sender without burst-mitigation
> > loses packets in the network due to the burst when a sender with
> > burst-mitigation does not. So, I agree that there is a tradeoff. And
> > that we should write it out clearly.
>=20
> Well, this tradeoff is fairly obvious I think. You could include for
> completeness I suppose. I'm not too concerned about this topic, because
> we always face this tradeoff when dealing with correct congestion
> control. Correct congestion control almost always trades off a sender's
> throughput with the network's health. So, if sender's wanted to be
> greedy, they COULD avoid congestion control altogether. However, we
> don't see this as a default behavior in major operating systems.

Well, this is an experimental modification that makes TCP sender
behaviour more conservative than "correct congestion control" by
reducing cwnd after FlightSize drops suddenly. So it might be worthwhile
to think about possible scenarios for a while, to figure out if there
are cases (due to network behaviour, or due to behaviour of the other
end) where this approach could be unnecessarily disadvantageous compared
to the standard and "legal" TCP flows. Buggy receivers was not the best
of examples, but I was looking for a possible case where FlightSize
would be frequently reduced by more than BLimit for some reason, and
thereby TCP sender would not be able to increase its cwnd in a normal
way. Some nonconventional SACK flavor, or a buggy delayed ack
implementation is obviously a problem that receiver itself should fix,
but on the other hand, at the same time your neighbour might be able to
deliver the data effectively just by following the standard congestion
control rules.

The other side of the trade-off is that it is unknown how harmful
transient micro-bursts are, and whether they are better dealt with
Active Queue Management, Rate-based pacing, or some other mechanism.

- Pasi


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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 TCP Maintenance and Minor Extensions Working Group of the IETF.

	Title		: Improving the Robustness of TCP to Non-Congestion Events
	Author(s)	: S. Bhandarkar, et al.
	Filename	: draft-ietf-tcpm-tcp-dcr-07.txt
	Pages		: 18
	Date		: 2006-2-15
	
This document specifies Non-Congestion Robustness (NCR) for TCP.  In
the absence of explicit congestion notification from the network TCP
uses loss as an indication of congestion.  One of the ways TCP
detects loss is using the arrival of three duplicate acknowledgments.
However, this heuristic is not always correct, notably in the case
when network paths reorder segments (for whatever reason), resulting
in degraded performance.  TCP-NCR is designed to mitigate this
degraded performance by increasing the number of duplicate
acknowledgments required to trigger loss recovery, based on the
current state of the connection, in an effort to better disambiguate
true segment loss from segment reordering.  This document specifies
the changes to TCP, as well as the costs and benefits of these
modifications.

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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 TCP Maintenance and Minor Extensions Working Group of the IETF.

	Title		: Improving TCP's Robustness to Blind In-Window Attacks
	Author(s)	: R. Stewart, M. Dalal
	Filename	: draft-ietf-tcpm-tcpsecure-04.txt
	Pages		: 27
	Date		: 2006-2-15
	
A recent study indicates that some types of TCP connections have an
   increased vulnerability to spoofed packet injection attacks than
   previously believed [SITW].  TCP has historically been considered
   protected against spoofed packet injection attacks by relying on the
   fact that it is difficult to guess the 4-tuple (the source and
   destination IP addresses and the source and destination ports) in
   combination with the 32 bit sequence number(s).  A combination of
   increasing window sizes and applications using a longer term
   connections (e.g.  H-323 or Border Gateway Protocol [RFC1771]) have
   left modern TCP implementation more vulnerable to these types of
   spoofed packet injection attacks.

   Note: Both [SITW] and [DTASA] provide charts which can give the
   reader an idea as to the time it takes to penetrate an unprotected
   system.

   Many of these long term TCP applications tend to have predictable IP
   addresses and ports which makes it far easier for the 4-tuple to be
   guessed.  Having guessed the 4-tuple correctly, an attacker can
   inject a RST, SYN or DATA segment into a TCP connection by carefly
   crafting the sequence number of the spoofed segment to be in the
   current receive window.  This can cause the connection to either
   abort or possibly cause data corruption.  This document proposes
   small modifications to the way TCP handles inbound segments that can
   reduce the probability of such an attack.

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Internet-Drafts@ietf.org wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
> This draft is a work item of the TCP Maintenance and Minor Extensions Working Group of the IETF.
> 
> 	Title		: Improving TCP's Robustness to Blind In-Window Attacks
> 	Author(s)	: R. Stewart, M. Dalal
> 	Filename	: draft-ietf-tcpm-tcpsecure-04.txt
> 	Pages		: 27
> 	Date		: 2006-2-15

Some of the document is incomplete; the ACKs section in particular is
cutoff in mid-sentence.

Also, the pointer to my ID is very old (it's been at -02 since Oct
2005), and the reference to Watson is incomplete (this was pointed out
before).

Any chance of posting a correction to these (the former might indicate
other text problems throughout) very shortly?

Joe
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From tcpm-bounces@ietf.org Thu Feb 16 09:51:51 2006
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To: "Duke, Martin" <Martin.Duke@boeing.com>
From: Mark Allman <mallman@icir.org>
Subject: Re: [tcpm] RFC2581bis 
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Folks-

> I have concerns about setting ssthresh after multiple timeouts
> (Equation 5).  My initial feeling is that cutting ssthresh in half
> each time is too conservative.
>
> [...]
> 
> So as an alternative, perhaps we should leave ssthresh unchanged after
> the first retransmit.

Vern, Ethan and I have been chatting about this proposal.  We're a
little bit torn here because:

  * Our intent was to revise RFC 2581 for two reasons: (1) to roll in
    small changes to TCP that are already standards track (e.g., limited
    transmit and the larger initial window) and (2) to fix bugs
    identified in 2581.  It seems to us that the above is outside the
    scope of either of these tasks.

  * It seems to the three of us that setting ssthresh on the first RTO
    and then not adjusting it on subsequent RTOs for the same sequence
    number is safe and may well be a reasonable path.

So, how to reconcile these things?  Do we really want to require Martin
(or someone) to write a standards track RFC for something so small such
that it can someday be rolled into the main spec?  That seems like a
pretty high price.

So, we're asking the WG whether (a) the change seems reasonable and (b)
whether it is OK to include this small tweak into the current revision
of RFC2581.

(Note: (b) is being asked about this small change only.  We still think
that (1) and (2) should be the guiding principles for the revision.
Although, of course, the WG can decide otherwise, as well.)

Thanks!

allman



-- 
Mark Allman -- ICIR/ICSI -- http://www.icir.org/mallman/




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Subject: RE: [tcpm] RFC2581bis 
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A) I agree with Martin. It is too conservative and overkills the
throughput of the connection, especially in mobile cellular systems with
which I am more familiar and I can say that given the high RTT (and
RTO!) the impact on throughput is quite dramatic.

Do we need to be a bit more precise? For example, what about a timeout
during the loss recovery phase? At the time of the fast retransmit
ssthresh is set to half the value of cwnd. But if before the end of the
loss recovery phase there is a timeout then ssthresh is halved again as
I understand it from RFC 2581. My opinion is that this is also too
conservative.

B) My opinion is that such small change fits very appropriately into RFC
2581.

Jose

>Folks-

> I have concerns about setting ssthresh after multiple timeouts
> (Equation 5).  My initial feeling is that cutting ssthresh in half
> each time is too conservative.
>
> [...]
>=20
> So as an alternative, perhaps we should leave ssthresh unchanged after
> the first retransmit.

Vern, Ethan and I have been chatting about this proposal.  We're a
little bit torn here because:

  * Our intent was to revise RFC 2581 for two reasons: (1) to roll in
    small changes to TCP that are already standards track (e.g., limited
    transmit and the larger initial window) and (2) to fix bugs
    identified in 2581.  It seems to us that the above is outside the
    scope of either of these tasks.

  * It seems to the three of us that setting ssthresh on the first RTO
    and then not adjusting it on subsequent RTOs for the same sequence
    number is safe and may well be a reasonable path.

So, how to reconcile these things?  Do we really want to require Martin
(or someone) to write a standards track RFC for something so small such
that it can someday be rolled into the main spec?  That seems like a
pretty high price.

So, we're asking the WG whether (a) the change seems reasonable and (b)
whether it is OK to include this small tweak into the current revision
of RFC2581.

(Note: (b) is being asked about this small change only.  We still think
that (1) and (2) should be the guiding principles for the revision.
Although, of course, the WG can decide otherwise, as well.)

Thanks!

allman=20

-----Original Message-----
From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf Of
Mark Allman
Sent: 16 February 2006 14:52
To: Duke, Martin
Cc: Ethan Blanton; tcpm@ietf.org; vern@icir.org
Subject: Re: [tcpm] RFC2581bis=20

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From tcpm-bounces@ietf.org Thu Feb 16 10:43:52 2006
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Subject: Re: [tcpm] RFC2581bis 
Date: Thu, 16 Feb 2006 16:46:02 +0100
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Hi all,

I have no opinion about A) (i.e. I'd rather leave this to people who
have some operational experience with this matter), but agree about B)
(fits in the 2581 update).

Cheers,
Michael


----- Original Message ----- 
From: "Gil Jose-josegil1" <josegil1@motorola.com>
To: <mallman@icir.org>; "Duke, Martin" <Martin.Duke@boeing.com>
Cc: "Ethan Blanton" <eblanton@cs.ohiou.edu>; <tcpm@ietf.org>;
<vern@icir.org>
Sent: Thursday, February 16, 2006 4:26 PM
Subject: RE: [tcpm] RFC2581bis


A) I agree with Martin. It is too conservative and overkills the
throughput of the connection, especially in mobile cellular systems with
which I am more familiar and I can say that given the high RTT (and
RTO!) the impact on throughput is quite dramatic.

Do we need to be a bit more precise? For example, what about a timeout
during the loss recovery phase? At the time of the fast retransmit
ssthresh is set to half the value of cwnd. But if before the end of the
loss recovery phase there is a timeout then ssthresh is halved again as
I understand it from RFC 2581. My opinion is that this is also too
conservative.

B) My opinion is that such small change fits very appropriately into RFC
2581.

Jose

>Folks-

> I have concerns about setting ssthresh after multiple timeouts
> (Equation 5).  My initial feeling is that cutting ssthresh in half
> each time is too conservative.
>
> [...]
>
> So as an alternative, perhaps we should leave ssthresh unchanged after
> the first retransmit.

Vern, Ethan and I have been chatting about this proposal.  We're a
little bit torn here because:

  * Our intent was to revise RFC 2581 for two reasons: (1) to roll in
    small changes to TCP that are already standards track (e.g., limited
    transmit and the larger initial window) and (2) to fix bugs
    identified in 2581.  It seems to us that the above is outside the
    scope of either of these tasks.

  * It seems to the three of us that setting ssthresh on the first RTO
    and then not adjusting it on subsequent RTOs for the same sequence
    number is safe and may well be a reasonable path.

So, how to reconcile these things?  Do we really want to require Martin
(or someone) to write a standards track RFC for something so small such
that it can someday be rolled into the main spec?  That seems like a
pretty high price.

So, we're asking the WG whether (a) the change seems reasonable and (b)
whether it is OK to include this small tweak into the current revision
of RFC2581.

(Note: (b) is being asked about this small change only.  We still think
that (1) and (2) should be the guiding principles for the revision.
Although, of course, the WG can decide otherwise, as well.)

Thanks!

allman

-----Original Message-----
From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On Behalf Of
Mark Allman
Sent: 16 February 2006 14:52
To: Duke, Martin
Cc: Ethan Blanton; tcpm@ietf.org; vern@icir.org
Subject: Re: [tcpm] RFC2581bis

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From tcpm-bounces@ietf.org Thu Feb 16 12:04:06 2006
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Subject: RE: [tcpm] RFC2581bis 
Date: Thu, 16 Feb 2006 09:03:52 -0800
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Thread-Topic: [tcpm] RFC2581bis 
Thread-Index: AcYzCNaIfjLAFilrSPa2ZJooW+AnsAACfuhQ
From: "Anantha Ramaiah \(ananth\)" <ananth@cisco.com>
To: <mallman@icir.org>, "Duke, Martin" <Martin.Duke@boeing.com>
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Regarding a) regarding the larger initial window, I think it makes sense
to have it with RFC 2581. Have no opinion on the limited transmit making
its reference in RFC 2581.

Also I would think it is probably ok to go with option b) ie update the
existing 2581. Since the changes doesn't involve a major re-write, only
some handfull of changes..(though important)

-Anantha
> -----Original Message-----
> From: tcpm-bounces@ietf.org [mailto:tcpm-bounces@ietf.org] On=20
> Behalf Of Mark Allman
> Sent: Thursday, February 16, 2006 6:52 AM
> To: Duke, Martin
> Cc: Ethan Blanton; tcpm@ietf.org; vern@icir.org
> Subject: Re: [tcpm] RFC2581bis=20
>=20
>=20
> Folks-
>=20
> > I have concerns about setting ssthresh after multiple timeouts=20
> > (Equation 5).  My initial feeling is that cutting ssthresh in half=20
> > each time is too conservative.
> >
> > [...]
> >=20
> > So as an alternative, perhaps we should leave ssthresh=20
> unchanged after=20
> > the first retransmit.
>=20
> Vern, Ethan and I have been chatting about this proposal. =20
> We're a little bit torn here because:
>=20
>   * Our intent was to revise RFC 2581 for two reasons: (1) to roll in
>     small changes to TCP that are already standards track=20
> (e.g., limited
>     transmit and the larger initial window) and (2) to fix bugs
>     identified in 2581.  It seems to us that the above is outside the
>     scope of either of these tasks.
>=20
>   * It seems to the three of us that setting ssthresh on the first RTO
>     and then not adjusting it on subsequent RTOs for the same sequence
>     number is safe and may well be a reasonable path.
>=20
> So, how to reconcile these things?  Do we really want to=20
> require Martin (or someone) to write a standards track RFC=20
> for something so small such that it can someday be rolled=20
> into the main spec?  That seems like a pretty high price.
>=20
> So, we're asking the WG whether (a) the change seems=20
> reasonable and (b) whether it is OK to include this small=20
> tweak into the current revision of RFC2581.
>=20
> (Note: (b) is being asked about this small change only.  We=20
> still think that (1) and (2) should be the guiding principles=20
> for the revision.
> Although, of course, the WG can decide otherwise, as well.)
>=20
> Thanks!
>=20
> allman
>=20
>=20
>=20
> --
> Mark Allman -- ICIR/ICSI -- http://www.icir.org/mallman/
>=20
>=20
>=20
>=20

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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

I agree with Mark:

	- the modification needed is beyond the scope of 2581bis

The change should be documented as either proposed standard or
experimental (the latter justified since it's out in the wild anyway),
at which point 2581bis could cite it, but coupling these two documents
(one in-hand, one not even drafted yet) would stall 2581bis far too long.

There will always be mods to TCP that occur after any particular
document revision; let's not hold one up over the other.

Joe


Mark Allman wrote:
> Folks-
> 
> 
>>I have concerns about setting ssthresh after multiple timeouts
>>(Equation 5).  My initial feeling is that cutting ssthresh in half
>>each time is too conservative.
>>
>>[...]
>>
>>So as an alternative, perhaps we should leave ssthresh unchanged after
>>the first retransmit.
> 
> 
> Vern, Ethan and I have been chatting about this proposal.  We're a
> little bit torn here because:
> 
>   * Our intent was to revise RFC 2581 for two reasons: (1) to roll in
>     small changes to TCP that are already standards track (e.g., limited
>     transmit and the larger initial window) and (2) to fix bugs
>     identified in 2581.  It seems to us that the above is outside the
>     scope of either of these tasks.
> 
>   * It seems to the three of us that setting ssthresh on the first RTO
>     and then not adjusting it on subsequent RTOs for the same sequence
>     number is safe and may well be a reasonable path.
> 
> So, how to reconcile these things?  Do we really want to require Martin
> (or someone) to write a standards track RFC for something so small such
> that it can someday be rolled into the main spec?  That seems like a
> pretty high price.
> 
> So, we're asking the WG whether (a) the change seems reasonable and (b)
> whether it is OK to include this small tweak into the current revision
> of RFC2581.
> 
> (Note: (b) is being asked about this small change only.  We still think
> that (1) and (2) should be the guiding principles for the revision.
> Although, of course, the WG can decide otherwise, as well.)
> 
> Thanks!
> 
> allman
> 
> 
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www1.ietf.org/mailman/listinfo/tcpm
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I agree with Joe. Not only would this change stall the 2581bis, but it
would create a precedent that we might regret later. New ideas,
regardless of how small they may seem, should follow the normal course
of events. There will most likely be a 2581bis-bis later on down the road.

-- 
Armando
www.armandocaro.net

Joe Touch wrote:
> I agree with Mark:
> 
> 	- the modification needed is beyond the scope of 2581bis
> 
> The change should be documented as either proposed standard or
> experimental (the latter justified since it's out in the wild anyway),
> at which point 2581bis could cite it, but coupling these two documents
> (one in-hand, one not even drafted yet) would stall 2581bis far too long.
> 
> There will always be mods to TCP that occur after any particular
> document revision; let's not hold one up over the other.
> 
> Joe

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From: David Borman <dab@weston.borman.com>
Subject: Re: [tcpm] RFC2581bis 
Date: Thu, 16 Feb 2006 15:19:04 -0600
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How about adding it as an appendix or a discussion item?  That would  
document the issue and the proposed solution, but defer making it  
part of the standard.

			-David Borman

On Feb 16, 2006, at 8:51 AM, Mark Allman wrote:

>
> Folks-
>
>> I have concerns about setting ssthresh after multiple timeouts
>> (Equation 5).  My initial feeling is that cutting ssthresh in half
>> each time is too conservative.
>>
>> [...]
>>
>> So as an alternative, perhaps we should leave ssthresh unchanged  
>> after
>> the first retransmit.
>
> Vern, Ethan and I have been chatting about this proposal.  We're a
> little bit torn here because:
>
>   * Our intent was to revise RFC 2581 for two reasons: (1) to roll in
>     small changes to TCP that are already standards track (e.g.,  
> limited
>     transmit and the larger initial window) and (2) to fix bugs
>     identified in 2581.  It seems to us that the above is outside the
>     scope of either of these tasks.
>
>   * It seems to the three of us that setting ssthresh on the first RTO
>     and then not adjusting it on subsequent RTOs for the same sequence
>     number is safe and may well be a reasonable path.
>
> So, how to reconcile these things?  Do we really want to require  
> Martin
> (or someone) to write a standards track RFC for something so small  
> such
> that it can someday be rolled into the main spec?  That seems like a
> pretty high price.
>
> So, we're asking the WG whether (a) the change seems reasonable and  
> (b)
> whether it is OK to include this small tweak into the current revision
> of RFC2581.
>
> (Note: (b) is being asked about this small change only.  We still  
> think
> that (1) and (2) should be the guiding principles for the revision.
> Although, of course, the WG can decide otherwise, as well.)
>
> Thanks!
>
> allman
>
>
>
> -- 
> Mark Allman -- ICIR/ICSI -- http://www.icir.org/mallman/
>
>
>
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www1.ietf.org/mailman/listinfo/tcpm


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David Borman wrote:
> How about adding it as an appendix or a discussion item?  That would 
> document the issue and the proposed solution, but defer making it  part
> of the standard.
> 
>             -David Borman

It could/should be listed as an open issue, under "pending mods that are
not covered by this document".

(I'd avoid an appendix devoted to this issue; IMO, the issue should be
covered elsewhere in detail, not in this doc).

Joe

> 
> On Feb 16, 2006, at 8:51 AM, Mark Allman wrote:
> 
>>
>> Folks-
>>
>>> I have concerns about setting ssthresh after multiple timeouts
>>> (Equation 5).  My initial feeling is that cutting ssthresh in half
>>> each time is too conservative.
>>>
>>> [...]
>>>
>>> So as an alternative, perhaps we should leave ssthresh unchanged  after
>>> the first retransmit.
>>
>>
>> Vern, Ethan and I have been chatting about this proposal.  We're a
>> little bit torn here because:
>>
>>   * Our intent was to revise RFC 2581 for two reasons: (1) to roll in
>>     small changes to TCP that are already standards track (e.g.,  limited
>>     transmit and the larger initial window) and (2) to fix bugs
>>     identified in 2581.  It seems to us that the above is outside the
>>     scope of either of these tasks.
>>
>>   * It seems to the three of us that setting ssthresh on the first RTO
>>     and then not adjusting it on subsequent RTOs for the same sequence
>>     number is safe and may well be a reasonable path.
>>
>> So, how to reconcile these things?  Do we really want to require  Martin
>> (or someone) to write a standards track RFC for something so small  such
>> that it can someday be rolled into the main spec?  That seems like a
>> pretty high price.
>>
>> So, we're asking the WG whether (a) the change seems reasonable and  (b)
>> whether it is OK to include this small tweak into the current revision
>> of RFC2581.
>>
>> (Note: (b) is being asked about this small change only.  We still  think
>> that (1) and (2) should be the guiding principles for the revision.
>> Although, of course, the WG can decide otherwise, as well.)
>>
>> Thanks!
>>
>> allman
>>
>>
>>
>> -- 
>> Mark Allman -- ICIR/ICSI -- http://www.icir.org/mallman/
>>
>>
>>
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www1.ietf.org/mailman/listinfo/tcpm
> 
> 
> 
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www1.ietf.org/mailman/listinfo/tcpm
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Subject: RE: [tcpm] RFC2581bis
Date: Fri, 17 Feb 2006 08:08:58 -0800
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Thread-Topic: [tcpm] RFC2581bis
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From: "Duke, Martin" <Martin.Duke@boeing.com>
To: "Joe Touch" <touch@ISI.EDU>, "David Borman" <dab@weston.borman.com>
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I'm not sure I agree that this is outside the scope of the task at hand,
largely because I don't think that this is a change in what 2581
specifies.=20

I believe that the current specification can be interpreted two ways,
and in fact, Linux and BSD interpret them differently.  If we're going
to clarify the intent, as the last draft of 2581bis seems to do, I think
we should clarify it in the way that best prepares TCP to function in
the future with minimum modification.

Excerpt from the text of both documents is below.

Martin

(1) Original 2581 text:
"When a TCP sender detects segment loss using the retransmission
   timer, the value of ssthresh MUST be set to no more than the value
   given in equation 3:

      ssthresh =3D max (FlightSize / 2, 2*SMSS)            (3)

   As discussed above, FlightSize is the amount of outstanding data in
   the network.

   Implementation Note: an easy mistake to make is to simply use cwnd,
   rather than FlightSize, which in some implementations may
   incidentally increase well beyond rwnd."

Flightsize is described as:
"FLIGHT SIZE:  The amount of data that has been sent but not yet
      acknowledged."
But does that include data that has timed out?

(2) 2581bis adds the following explanatory text:
"On the other hand, when a TCP sender detects segment loss using the
    retransmission timer and the given segment has already been
    retransmitted at least once, the value of ssthresh MUST be set to no
    more than the value given in equation 5:

        ssthresh =3D max (ssthresh / 2, 2*SMSS)              (5)

    In other words, upon the first retransmission of a segment the value
    of ssthresh should be set to half the amount of outstanding data in
    the network, whereas on subsequent retransmissions the value of
    ssthresh should simply be halved."

-----Original Message-----
From: Joe Touch [mailto:touch@ISI.EDU]=20
Sent: Thursday, February 16, 2006 1:27 PM
To: David Borman
Cc: Ethan Blanton; tcpm@ietf.org; vern@icir.org; mallman@icir.org
Subject: Re: [tcpm] RFC2581bis

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1



David Borman wrote:
> How about adding it as an appendix or a discussion item?  That would=20
> document the issue and the proposed solution, but defer making it =20
> part of the standard.
>=20
>             -David Borman

It could/should be listed as an open issue, under "pending mods that are
not covered by this document".

(I'd avoid an appendix devoted to this issue; IMO, the issue should be
covered elsewhere in detail, not in this doc).

Joe

>=20
> On Feb 16, 2006, at 8:51 AM, Mark Allman wrote:
>=20
>>
>> Folks-
>>
>>> I have concerns about setting ssthresh after multiple timeouts=20
>>> (Equation 5).  My initial feeling is that cutting ssthresh in half=20
>>> each time is too conservative.
>>>
>>> [...]
>>>
>>> So as an alternative, perhaps we should leave ssthresh unchanged =20
>>> after the first retransmit.
>>
>>
>> Vern, Ethan and I have been chatting about this proposal.  We're a=20
>> little bit torn here because:
>>
>>   * Our intent was to revise RFC 2581 for two reasons: (1) to roll in
>>     small changes to TCP that are already standards track (e.g.,
limited
>>     transmit and the larger initial window) and (2) to fix bugs
>>     identified in 2581.  It seems to us that the above is outside the
>>     scope of either of these tasks.
>>
>>   * It seems to the three of us that setting ssthresh on the first
RTO
>>     and then not adjusting it on subsequent RTOs for the same
sequence
>>     number is safe and may well be a reasonable path.
>>
>> So, how to reconcile these things?  Do we really want to require =20
>> Martin (or someone) to write a standards track RFC for something so=20
>> small  such that it can someday be rolled into the main spec?  That=20
>> seems like a pretty high price.
>>
>> So, we're asking the WG whether (a) the change seems reasonable and =20
>> (b) whether it is OK to include this small tweak into the current=20
>> revision of RFC2581.
>>
>> (Note: (b) is being asked about this small change only.  We still =20
>> think that (1) and (2) should be the guiding principles for the
revision.
>> Although, of course, the WG can decide otherwise, as well.)
>>
>> Thanks!
>>
>> allman
>>
>>
>>
>> --
>> Mark Allman -- ICIR/ICSI -- http://www.icir.org/mallman/
>>
>>
>>
>> _______________________________________________
>> tcpm mailing list
>> tcpm@ietf.org
>> https://www1.ietf.org/mailman/listinfo/tcpm
>=20
>=20
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www1.ietf.org/mailman/listinfo/tcpm
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Duke, Martin wrote:
> I'm not sure I agree that this is outside the scope of the task at hand=
,
> largely because I don't think that this is a change in what 2581
> specifies.=20
>=20
> I believe that the current specification can be interpreted two ways,
> and in fact, Linux and BSD interpret them differently.  If we're going
> to clarify the intent, as the last draft of 2581bis seems to do, I thin=
k
> we should clarify it in the way that best prepares TCP to function in
> the future with minimum modification.
>=20
> Excerpt from the text of both documents is below.
>=20
> Martin
>=20
> (1) Original 2581 text:
> "When a TCP sender detects segment loss using the retransmission
>    timer, the value of ssthresh MUST be set to no more than the value
>    given in equation 3:
>=20
>       ssthresh =3D max (FlightSize / 2, 2*SMSS)            (3)
>=20
>    As discussed above, FlightSize is the amount of outstanding data in
>    the network.
>=20
>    Implementation Note: an easy mistake to make is to simply use cwnd,
>    rather than FlightSize, which in some implementations may
>    incidentally increase well beyond rwnd."
>=20
> Flightsize is described as:
> "FLIGHT SIZE:  The amount of data that has been sent but not yet
>       acknowledged."
> But does that include data that has timed out?

Send but timed out is still sent. You can't 'unsend' it. It can still
get there - and sometimes does.

Trying to reinterpret that recommendation to accommodate what *might*
happen in *some* contexts is speculation.

Joe


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Exactly.  I interpret the definition of FlightSize the same way.  If
FlightSize does not decrease with consecutive retransmission timeouts,
then ssthresh will only be halved after the first consecutive timeout.

That's the behavior that I advocate, which is NOT the behavior in
2581bis.=20

Martin

> -----Original Message-----
> From: Joe Touch [mailto:touch@ISI.EDU]=20
> Sent: Friday, February 17, 2006 8:37 AM
> To: Duke, Martin
Duke, Martin wrote:

<snip>

> >=20
> >       ssthresh =3D max (FlightSize / 2, 2*SMSS)            (3)
> >=20
> >    As discussed above, FlightSize is the amount of=20
> > outstanding data in the network.

<snip>

> > Flightsize is described as:
> > "FLIGHT SIZE:  The amount of data that has been sent but not yet
> >       acknowledged."
> > But does that include data that has timed out?
>=20
> Send but timed out is still sent. You can't 'unsend' it. It=20
> can still get there - and sometimes does.
>=20
> Trying to reinterpret that recommendation to accommodate what=20
> *might* happen in *some* contexts is speculation.
>=20
> Joe
>=20

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Hi, all,

Sorry for the long delay; I had been incorporating Ted's extensive
comments, and had not noted how these were handled yet. An update (03)
has been submitted with those and the following mods addressed as noted
below:

Pekka Savola wrote:
> My comments below.  There are still some things to fix, but most of the
> below should be straightforward.
> 
> substantial
> -----------
> 
> 1) the receiver buffer sizing assumption does not seem realistic in
> sufficiently many cases so that I think it should not be made.
...
> I just checked; Cisco has by default around 4 KB for BGP (for others,
> 32KB),
> and it isn't even possible to config it up to more than 128K.  Juniper
> has by default 16KB, and AFAICS there is no way to configure it.
> 
> ....  Maybe what needs to be done
> is calculate two tables: one where the buffers are adjusted so that the
> bandwidth can be used in a maximum manner and where the buffers are capped
> at (say) 16, 32 or 64 kB. That would give realistic calculations in
> either case without having to
> make assumptions about the buffers.

Done.

> (This same is in the first sentence of section 3.1.5.)
> 
> 2) I think the address filtering (section 3.2.1) is still a bit lacking.

Updated.

>  b) TTL checking problem with tunnels is still inaccurate. 

Updated.

>  c) At the beginning of section 3.2, you say "filtering is an indirect
> system which relies on other parties to filter packets ..." which is
> correct in the general case but not when examining the sessions inside a
> border (where you can _yourself_ do the filtering, without relying on
> anyone
> else).  This distinction is important depending on what kind of sessions
> you're interested in protecting and how large area your borders cover.

The text has been updated to clarify its original intent. I disagree
that "you can _yourself_ do the filting, without relying on anyone
else", except in trivial multiconnected stub network cases.


> 3) the processing complexity comparison between transport vs network
> solutions seems assumptive.

The section where this is mentioned talks about the processing
comparison of network and transport authentication; both are forms of
authentication with similar computational complexity. That's why this
section focuses on the other issues, such as demultiplexing and storage.

> 4) network layer solution may not be able to cope with on-path ICMP.

Section 4 was updated to clarify this point.

> 5) BTNS applicability is slightly too simplistic
> 
>    Alternative mechanisms are under development to address this
>    limitation, to allow publicly-accessible servers to secure
>    connections to clients not known in advance, or to allow unilateral
>    relaxation of identity validation so that the remaining protections
>    of IPsec to be available [38][39]. In particular, these mechanisms
>    can prevent a client (but without knowing who that client is) from
>    being affected by spoofing from other clients, even when the
>    attackers are on the communications path.
> 
> ==> AFAICS, "are on the communications path" is assumptive or simplistic.
> Either you must
> have a previously established relation (which you know is MITM-free) to
> a peer, or you meant something like "would later, after establishing
> the session, get on the communications path".  After all, BTNS does not
> protect from MITM if the MITM was there when the session was set up and
> you don't have prior knowledge of the peer.

The text above says "without knowing who the client is". It isn't
meaningful to talk about MITM in those cases - the MITM is the client at
that point. IMO, the text already has the sufficient caveat, and
sufficiently explains the protection provided.

> 6) The description why IPsec is not used to protect router-to-router
> communications fails to address the added IPsec processing requirements,
> which are typically prohibitive.

Updated earlier in sec 5.2 as to computational complexity.

>    IPsec, although widely available both in commercial routers and
>    commodity end-systems, is not often utilized except between parties
>    that already have a preexisting relationship (employee/employer,
>    between two ISPs, etc.) Servers to anonymous clients (e.g., customer/
>    business) or more open services (e.g., BGP, where routers may have
>    large numbers of peers) are unmanageable, due to the breadth and flux
>    of CAs. [...]
> 
> ==> This is too simplistic description of why IPsec isn't used between
> routers.  While management of certificates is a major issue, at least as
> important for me as an operator are the processing requirements of
> IPsec.  If you don't have dedicated hardware for the task, I'd rather spend
> the routing engine's CPU cycles on processing the BGP update packets (and
> other control data) rather than encryption, decryption, hashing, etc.

This paragraph is talking about key distribution; it is the earlier one
that talks about computation.

> Let me say this clearly: BTNS/Anonsec will not be used by routers unless
> they have dedicated IPsec hardware (and the vast majority of them won't in
> the short term at least).  (Hopefully this will be appropriately
> reflected in the  BTNS problem statement as well.)

Dedicated IPsec hardware is very cheap (e.g., for 100Mbps streams), and
IMO should be added to routers to protect their management CPUs from
computational overload. We have another draft in progress to talk about
reducing the CPU overload of IPsec, esp. for DOS attacks, but that was
already deemed out of scope for BTNS. I.e., computational effort was
taken off the table there during the BOF stages (despite my efforts to
the contrary ;-)

> 7) not sure if I agree with conclusions because we haven't yet reached
> consensus on the body yet (this is a placeholder :)

The document makes comparisons and recommendations in those contexts
throughout; I don't know if it's feasible to summarize them in a single
paragraph at the end.

...
> ==> not sure if I completely agree with these conclusions here (and
> in section 5), but when/if we get to agreement on the main body of the
> doc, it should be easier to get to consensus on the recommendations as
> well.

The recommendations have been there in several revisions with WG
opportunity to comment, and have been revised accordingly several times
already.

> editorial
> ---------
> 
>    This, coupled with the lack of sufficient address filtering to drop
>    such spoofed traffic, can increase the potential for off-path third-
>    party spoofing attacks.
> 
> ==> you include refs later in the doc for this, so maybe they should be
> here as well.

Done.

>    [...] even when not
>    fixed, there are only 65,384 valid source ports which may be
>    exhaustively attacked.
> 
> ==> how do you get that specific number?  Why this isn't 2^16 ?

Typo. Fixed as "approximately 65,000", which is sufficient for this doc.
I didn't want to get deep into which numbers cannot be used as source
ports (it seemed like unassigned ones shouldn't, but I couldn't verify
that).

>    protocol, although emerging as standard practice, unnecessarily
>    complicates the design and implementation of new protocols [28]  As
>    new attacks are discovered (SYN floods, RSTs, etc.), each protocol
> 
> ==> missing a period here.

Fixed.

>   In addition, many networks filter all ICMP packets because validation
>    may not be possible, especially because they can be injected from any
>    point on a path, and so cannot be selectively address filtered.
> 
> ==> actually, the problem with ICMP is that they can be injected from any
> point *at all*, not just restricted to the path, and regular address
> filtering doesn't help there (whereas address filtering could help with TCP
> RSTs in some cases).

Fixed.

>    Network solutions are computationally intensive and require pervasive
>    registration of certificate authorities with every possible endpoint.
>    This section explains these observations further.
> 
> ==> after the addition of address filtering in section 3.2, this needs
> to be
> updated to refer to just the ipsec part of network solutions.

Fixed.

>    4. the entire body of the packet is secured
> 
>    5. security associations are established only where identity is
>       authenticated by a know certificate authority or other pre-shared
>       key
> 
>    6. both on-path and off-path third-party spoofing attacks   must be
>       defeated
> 
> ==> these should be numbered from 1 to 3, as you're referring to (1) and
> (2)
> in the text below...

fixed.

>    [2]   Baker, F. and P. Savola, "Ingress Filtering for Multihomed
>          Networks," RFC 2827 / BCP 84, March 2004.
> 
>    [12]  Ferguson, P. and D. Senie, Network Ingress Ingress Filt
>          Defeating Denial of Service Attacks which employ IP Add
>          Spoofing," RFC 2267 / BCP 38, May 2000.
> 
> ==> BCP 84 is RFC 3704 while BCP 38 is 2827; earlier version of BCP 38 was
> 2267.

Fixed (thanks for the catch).

>    [38]  Touch, J., "ANONsec: Anonymous Security to Defend Again
>          Spoofing Attacks," draft-touch-anonsec-00 (work in prog
>          May 2004.
> 
>    [5]   Better Than Nothing Security [BTNS] BOF, IETF-61, Wash. DC.,
>          http://www.ietf.org/ietf/04nov/btns.txt
> 
> ==> these (and the text where they're referred to) have since been somewhat
> obsoleted.  Maybe it would be sufficient to refer to the latest BTNS WG
> docs and/or the BTNS charter?

Fixed.

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Sally,

(Resonse to: 
http://www1.ietf.org/mail-archive/web/tcpm/current/msg01671.html)

Thank you very much for spending time on this draft, and thank you for 
your comments! I am sorry that I am quite late in replying. Hope this 
email addresses your concerns, if not, then generates further discussion.


> * The network's tolerance of burstiness: [...]

I have added some text (based on your suggestions) that now seems more 
appropriate in the Discussion section (Section 4). I have also changed the 
tone somewhat... I do not stress the fact that increased deployment of AQM 
is happening (which I think was one of your points, correct me if I read 
wrong). The reason I changed the tone was to maintain focus on micro-burst 
mitigation, while mentioning the parallel effects of AQM. Let me know if 
you think the following captures it, or if you'd suggest mods.

     We emphasize that the question of whether or not micro-bursts need
     mitigation remains open. While use of micro-burst mitigation can
     reduce stress on network queues, congested routers may, in parallel,
     use some form of Active Queue Management (AQM) to better handle
     stress on network queues. Use of AQM in a router allows for better
     congestion control through better congestion signals (i.e., signals
     other than packet drops) and enables queues to accomodate transient
     bursts in the aggregate incoming traffic. Micro-burst mitigation may
     still provide some respite by reducing the amount of transient
     burstiness in the traffic aggregate at a queue.


> * Rate-based pacing:
> [...] I think it would be worth it to say that there are many cases 
> where rate-based pacing should be a big win over CWL, or over MaxBurst, 
> in terms of traffic dynamics. [...]

I have modified the para in the Related Work Section (Section 3) to 
reflect your suggestions. Also see the following argument.

       * Rate-Based Pacing [VH97] imposes a limitation on the rate of
         sending, and prevent bursts by pacing data into the network
         until the ACK clock is established.  Although this solution can
         be very effective in burst mitigation in some cases, it requires
         a new timer and parameters for pacing out the data segments.
         Further, as shown in [AB05], there are cases where there is no
         natural "lull" in the connection into which segments can be
         nicely paced, and as shown in [AST00], the tradeoffs involved in
         using pacing are not clear.  While the exact application of
         pacing clearly requires more research, there are several cases
         where pacing can be expected to mitigate bursts better than CWL
         (or MaxBurst or UI/LI). For instance, where CWL (also MaxBurst and
         UI/LI) will not be able to handle bursts caused by slow start,
         pacing could work well and prevent these transient
         bursts. Pacing can potentially also have a bigger impact on
         traffic aggregates, possibly significantly reducing burstiness in
         the aggregate at several timescales.

> E.g., in a short connection with only BLimit + 1 more packets to send, 
> but when only one ack will be arriving, due to ACK losses.

I do not use this example, because I am not sure how RBP will be in short 
connections, i.e., I am not clear on the tradeoffs here---for instance, 
what happens if the RTT value is seeded incorrectly in such a small 
connection? But I believe the larger point you wanted to convey has been 
captured in the para above. Let me know if not.

> But my suggestion would be to be more explicit about the potential 
> advantages of rate-based pacing, and to say something about it a little 
> earlier in the document.

I hope that para was able to capture it, but I'm not sure about saying 
something earlier in the document. This section is the first place that 
the other algorithms are mentioned and the pros and cons discussed. Do you 
think the entire section should appear earlier?


> * Security:
> I think that the security issues raised in Section 5 are pretty
> important, and could use more discussion.  This is another case
> where rate-based pacing would be more robust than CWL, I think.

I'll try to add more discussion on that point. Do you have anything 
specific in mind that you might want to suggest?

While I suspect that rate-based pacing may be more robust than CWL under 
ack loss, it is difficult for me to make a strong comparative statement 
without knowing what exactly would be used for rate-based pacing. 
Depending on the exact details, I suspect that rate-based pacing *could 
also possibly* be (differently) susceptible to issues such as ack loss. 
So, I am not sure that we should say anything about how the mentioned 
security issue applies to rate-based pacing.

> * Causes of micro-bursts: ACK-compression can also be a cause of 
> micro-bursts.  (That is, ACK-compression can cause micro-bursts as well 
> as macro-bursts.)

How so? I am thinking about it, but cannot figure out how that could be. 
A micro-burst is defined as a burst sent in response to one ACK. In time, 
I suppose ACK-compression could cause bursts at the same timescale. In 
other words, some macro-burst phenomena could occur at the timescales that 
the bursts resemble micro-bursts. Maybe we should state that explicitly in 
the discussion about macro-bursts. Something like

"Note that some macro-bursts could occur at the timescales of 
micro-bursts---for instance, back-to-back ACKs due to extreme 
ACK-compression could cause the consequent data burst to resemble a 
micro-burst from the network's point of view."

What do you think?


> * Section 2, step (1):
> "the only case where a micro-burst will not occur" ->
> "the only case where a micro-burst will not occur, if steps (2)
>  and (3) aren't used".

I am thinking of taking that entire sentence out---it does not add much 
value, but the cost of parsing it is high (and is getting higher). Unless 
someone sees enough value in it, that is.


> * "BLimit SHOULD be chosen such that bursts are no larger than those
> allowed by [RFC3390]."
>
> Why is this?  It is not obvious to me.

I was thinking about a fixed burst limit that one could apply for 
line-rate bursts, and it seemed reasonable to use 3390 as a basis. My 
(simplistic) reasoning was that since bursts from starting TCP connections 
should be absorbable by the network, 3390-defined bursts should be 
acceptable in general.

> For example, four-packet bursts, for 1500-byte packets, might be 
> perfectly acceptable, for a connection that has a sufficiently large 
> congestion window to send that many packets. And there is a cost to 
> BLimit being at most three 1500-byte packets.

I'm not sure I'm following. A large cwnd does not mean more segments can 
be sent (at line-rate) in response to _each ack_. BLimit is meant to 
impose that limit, not to limit the overall cwnd size. IMHO, having a 
large cwnd does not mean that a large burst is acceptable.

Yes, I agree that there is a cost to having BLimit at what it is. But is 
there a situation that demands larger line-rate bursts?

What do you think is a reasonable limit (static/dynamic)? I may not be 
following what you are saying; if so, let me know.


> * Maxburst:
> "An additional drawback of MaxBurst ...".
> What was the first drawback?  Introducing a separate control?  But
> the earlier sentence says that "using two different controls may
> make sense".

New text:
         However, a drawback with
         MaxBurst is that adding a second control brings with it the
         possibility of the two transmission controllers interacting
         poorly to cause undesirable side effects.


> "the two transmission controllers may interact poorly, causing
> undesirable side effects."
> Do you have any examples of this?  Or citations?  Or anything
> else to back this up?

Not really. The idea was to suggest that there is the possibility of poor 
interaction due to the introduction of a new control. Maybe the new text 
(see above) captures this more effectively?


> "When BLimit == MaxBurst, CWL and MaxBurst perform similarly [AB05]."
> I don't see how CWL and MaxBurst can perform similarly, and at
> the same time you can prefer one over the other.  I think it would
> be useful to look a little more closely for performance differences,
> or to say that there aren't any.

(I'm checking with my co-authors on this one. Will get back to you asap.)

> Depending on how it is implemented, MaxBurst could be made to work
> quite well with re-ordering, where each duplicate ack packet (possibly
> from reordering of data packets) could be responded-to with up to
> MaxBurst data packets (if allowed by the congestion window).
> (E.g., Aggressive MaxBurst from [AB05].)
> Can something like this be done with CWL as well?

Hmm. Offhand, I do not see how this can be done with CWL, but I will think 
about it more. This is a good point though. Maybe we should mention this 
in the related work section where we discuss pros and cons of MaxBurst.

Thanks again!
- jana

--------------------
Janardhan R. Iyengar
http://www.cis.udel.edu/~iyengar
Protocol Engineering Lab, University of Delaware


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The charter of the TCP Maintenance and Minor Extensions (tcpm) working group
in the Transport Area of the IETF has been updated. For additional
information, please contact the Area Directors or the working group Chairs.

+++

TCP Maintenance and Minor Extensions (tcpm)
=============================================

Current Status: Active Working Group 

Chair(s):
Ted Faber <faber@isi.edu> 
Mark Allman <mallman@icir.org> 

Transport Area Director(s):
Allison Mankin <mankin@psg.com> 
Jon Peterson <jon.peterson@neustar.biz> 

Transport Area Advisor:
Jon Peterson <jon.peterson@neustar.biz> 

Mailing Lists:
General Discussion: tcpm@ietf.org
To Subscribe: https://www.ietf.org/mailman/listinfo/tcpm
Archive: http://www.ietf.org/mail-archive/web/tcpm/index.html

Description of Working Group:

TCP is currently the Internet's predominant transport protocol. 
To maintain TCP's utility the IETF has regularly updated both 
the protocol itself and the congestion control algorithms 
implemented by the protocol that are crucial for the stability 
of the Internet. These changes reflect our evolving 
understanding of transport protocols, congestion control and new 
needs presented by an ever-changing network. The TCPM WG will 
provide a venue within the IETF to work on these issues. The WG 
will serve several purposes: 

* The WG will mostly focus on maintenance issues (e.g., bug 
fixes) and modest changes to the protocol and algorithms 
that maintain TCP's utility. 

* The WG will be a venue for moving current TCP specifications 
along the standards track (as community energy is available 
for such efforts). 

* The WG will write a document that outlines "what is TCP". 
This document will be a roadmap of sorts to the various TCP 
specifications in the RFC series. 

TCPM will take a subset of the work which has been conducted in 
the Transport Area WG over the past several years. 
Specifically, some of the WG's initial work will be moved from 
the Transport Area WG (tsvwg). 

TCPM is expected to be the working group within the IETF to 
handle TCP changes. Proposals for additional TCP work items 
should be brought up within the working group. While 
fundamental changes to TCP or its congestion control algorithms 
(e.g., departure from loss-based congestion control) should be 
brought through TCPM, it is expected that such large changes 
will ultimately be handled by the Transport Area WG (tsvwg). 
All additional work items for TCPM will, naturally, require the 
approval of the Transport Services Area Area Directors and the 
IESG. 

TCP's congestion control algorithms are the model followed by 
alternate transports (e.g., SCTP and (in some cases) DCCP). In 
addition, the IETF has recently worked on several documents 
about algorithms that are specified for multiple protocols 
(e.g., TCP and SCTP) in the same document. Which WG shepherds 
such documents in the future will determined on a case-by-case 
basis. In any case, the TCPM WG will remain in close contact 
with other relevant WGs working on these protocols to ensure 
openness and stringent review from all angles. 

Specific Goals: 

* A document specifying a way to share the local "User TimeOut" 
value with the peer such that TCP connections can withstand long 
periods of disconnection. 

* The WG is working on an experimental technique to add robustness 
to TCP against packet reordering having a negative impact on 
performance. 

* The WG is coming to grips with how to deal with spoofed segments 
that can tear down connections, cause data corruption or 
performance problems. To this end the WG is generating an 
overview document as well as a scheme that mitigates some of the 
spoofed segment issues using a challenge-response scheme to 
reduce the probabilities of a connection being impacted. 

* The WG is writing an informational document about the ways in 
which TCPs can handle ICMP "soft errors". 

* The WG is updating the specification for Explicit Congestion 
Notification to allow for the use of ECN during part of TCP's 
three-way handshake to aid performance for short transfers. 

Milestones: 

Apr 06 Submit NCR Reordering Mitigation draft to the IESG for 
publication as an Experimental RFC. 

Apr 06 Submit ECN-SYN document to the IESG for publication as a 
Proposed Standard RFC. 

May 06 Submit overview of spoofing attacks against TCP to IESG 
for publication as an Informational RFC. 

May 06 Submit In-Window Attack draft to IESG for publication as 
a Proposed Standard RFC. 

May 06 Submit User TimeOut option document to the IESG for 
publication as a Proposed Standard RFC. 

May 06 Submit soft errors document to the IESG for publication 
as an Informational RFC. 



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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 TCP Maintenance and Minor Extensions Working Group of the IETF.

	Title		: Defending TCP Against Spoofing Attacks
	Author(s)	: J. Touch
	Filename	: draft-ietf-tcpm-tcp-antispoof-03.txt
	Pages		: 27
	Date		: 2006-2-22
	
Recent analysis of potential attacks on core Internet infrastructure 
indicates an increased vulnerability of TCP connections to spurious 
resets (RSTs), sent with forged IP source addresses (spoofing).  TCP 
has always been susceptible to such RST spoofing attacks, which were 
indirectly protected by checking that the RST sequence number was 
inside the current receive window, as well as via the obfuscation of 
TCP endpoint and port numbers.  For pairs of well-known endpoints 
often over predictable port pairs, such as BGP or between web servers 
and well-known large-scale caches, increases in the path bandwidth-
delay product of a connection have sufficiently increased the receive 
window space that off-path third parties can guess a viable RST 
sequence number.  The susceptibility to attack increases as the 
square of the bandwidth, thus presents a significant vulnerability 
for recent high-speed networks.  This document addresses this 
vulnerability, discussing proposed solutions at the transport level 
and their inherent challenges, as well as existing network level 
solutions and the feasibility of their deployment.  This document 
focuses on vulnerabilities due to spoofed TCP segments, and includes 
a discussion of related ICMP spoofing attacks on TCP connections.

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>> * The network's tolerance of burstiness: [...]
>
>I have added some text (based on your suggestions) that now seems more 
>appropriate in the Discussion section (Section 4). I have also changed the 
>tone somewhat... I do not stress the fact that increased deployment of AQM 
>is happening (which I think was one of your points, correct me if I read 
>wrong). The reason I changed the tone was to maintain focus on micro-burst 
>mitigation, while mentioning the parallel effects of AQM. Let me know if 
>you think the following captures it, or if you'd suggest mods.
>
>     We emphasize that the question of whether or not micro-bursts need
>     mitigation remains open. While use of micro-burst mitigation can
>     reduce stress on network queues, congested routers may, in parallel,
>     use some form of Active Queue Management (AQM) to better handle
>     stress on network queues. Use of AQM in a router allows for better
>     congestion control through better congestion signals (i.e., signals
>     other than packet drops) and enables queues to accomodate transient
>     bursts in the aggregate incoming traffic. Micro-burst mitigation may
>     still provide some respite by reducing the amount of transient
>     burstiness in the traffic aggregate at a queue.

Yep, that sounds good to me.

>> * Rate-based pacing:
>> [...] I think it would be worth it to say that there are many cases 
>> where rate-based pacing should be a big win over CWL, or over MaxBurst, 
>> in terms of traffic dynamics. [...]
>
>I have modified the para in the Related Work Section (Section 3) to 
>reflect your suggestions. Also see the following argument.
>
>       * Rate-Based Pacing [VH97] imposes a limitation on the rate of
>         sending, and prevent bursts by pacing data into the network
         until the ACK clock is established.  Although this solution can
>         be very effective in burst mitigation in some cases, it requires
>         a new timer and parameters for pacing out the data segments.
>         Further, as shown in [AB05], there are cases where there is no
>         natural "lull" in the connection into which segments can be
>         nicely paced, and as shown in [AST00], the tradeoffs involved in
>         using pacing are not clear.  While the exact application of
>         pacing clearly requires more research, there are several cases
>         where pacing can be expected to mitigate bursts better than CWL
>         (or MaxBurst or UI/LI). For instance, where CWL (also MaxBurst and
>         UI/LI) will not be able to handle bursts caused by slow start,
>         pacing could work well and prevent these transient
>         bursts. Pacing can potentially also have a bigger impact on
>         traffic aggregates, possibly significantly reducing burstiness in
>         the aggregate at several timescales.
>
>> E.g., in a short connection with only BLimit + 1 more packets to send, 
>> but when only one ack will be arriving, due to ACK losses.
>
>I do not use this example, because I am not sure how RBP will be in short 
>connections, i.e., I am not clear on the tradeoffs here---for instance, 
>what happens if the RTT value is seeded incorrectly in such a small 
>connection? But I believe the larger point you wanted to convey has been 
>captured in the para above. Let me know if not.

That sounds fine to me.

>> But my suggestion would be to be more explicit about the potential 
>> advantages of rate-based pacing, and to say something about it a little 
>> earlier in the document.
>
>I hope that para was able to capture it, but I'm not sure about saying 
>something earlier in the document. This section is the first place that 
>the other algorithms are mentioned and the pros and cons discussed. Do you 
>think the entire section should appear earlier?

No, I think it is fine.


>> * Security:
>> I think that the security issues raised in Section 5 are pretty
>> important, and could use more discussion.  This is another case
>> where rate-based pacing would be more robust than CWL, I think.
>
>I'll try to add more discussion on that point. Do you have anything 
>specific in mind that you might want to suggest?

Nope, not really.

>While I suspect that rate-based pacing may be more robust than CWL under 
>ack loss, it is difficult for me to make a strong comparative statement 
>without knowing what exactly would be used for rate-based pacing. 
>Depending on the exact details, I suspect that rate-based pacing *could 
>also possibly* be (differently) susceptible to issues such as ack loss. 
>So, I am not sure that we should say anything about how the mentioned 
>security issue applies to rate-based pacing.

I disagree.  The point of this paper is not be be a used-car salesman
for CWL (sorry), the point is to evaluate CWL as fairly as you are able,
including considering its weaknesses, and considering when alternate
proposals might be more robust in some particular respects.  If you
"suspect that rate-based pacing may be more robust than CWL under
ack loss", or under reordering of data or acks, then I think it
would be good to think about it a bit, and to say something.  The
question isn't whether it would be *possible* to design a rate-based
pacing mechanism that would share the same weaknesses, but whether
it would be *possible* to design a rate-based pacing mechanism that
didn't.  In my opinion.  


>> * Causes of micro-bursts: ACK-compression can also be a cause of
>> micro-bursts.  (That is, ACK-compression can cause micro-bursts as well
>> as macro-bursts.)
>
>How so? I am thinking about it, but cannot figure out how that could be.
>A micro-burst is defined as a burst sent in response to one ACK. In time,
>I suppose ACK-compression could cause bursts at the same timescale. In
>other words, some macro-burst phenomena could occur at the timescales that
>the bursts resemble micro-bursts. Maybe we should state that explicitly in
>the discussion about macro-bursts. Something like
>
>"Note that some macro-bursts could occur at the timescales of
>micro-bursts---for instance, back-to-back ACKs due to extreme
>ACK-compression could cause the consequent data burst to resemble a
>micro-burst from the network's point of view."
>
>What do you think?

My advice would be to broaden your definition of micro-bursts to include
bursts on small time scales due to ACK compression.  Then you could
talk about which mechanisms do or don't ameliorate the bursts due
to ACK compression...


...
>> * "BLimit SHOULD be chosen such that bursts are no larger than those
>> allowed by [RFC3390]."
>>
>> Why is this?  It is not obvious to me.
>
>I was thinking about a fixed burst limit that one could apply for
>line-rate bursts, and it seemed reasonable to use 3390 as a basis. My
>(simplistic) reasoning was that since bursts from starting TCP connections
>should be absorbable by the network, 3390-defined bursts should be
>acceptable in general.

I might be worth talking about the potential costs of an
unnecessarily-small burst limit, so that it is clear that it is
a case of making a tradeoff somewhere.  

>> For example, four-packet bursts, for 1500-byte packets, might be
>> perfectly acceptable, for a connection that has a sufficiently large
>> congestion window to send that many packets. And there is a cost to
>> BLimit being at most three 1500-byte packets.
>
>I'm not sure I'm following. A large cwnd does not mean more segments can
>be sent (at line-rate) in response to _each ack_. BLimit is meant to
>impose that limit, not to limit the overall cwnd size. IMHO, having a
>large cwnd does not mean that a large burst is acceptable.

When you say that "BLimit SHOULD be chosen such that bursts are no larger
than those allowed by [RFC3390]", you are saying that
BLimit should not allow micro-bursts of four 1500-byte packets, but
should only allow micro-bursts of three 1500-byte packets, right?
I am just asking what is wrong with a micro-burst of four 1500-byte
packets?  (For a connection that has a congestion window large
enough to send four 1500-byte packets.)  I understand that BLimit
doesn't limit the overall cwnd size.
 
>Yes, I agree that there is a cost to having BLimit at what it is. But is
>there a situation that demands larger line-rate bursts?

Yep, consider a connection with only four 1500-byte packets left
to send, but only one ACK coming in (because delayed ACKs are used,
and one of the last two ACKs was dropped or reordered).  

>What do you think is a reasonable limit (static/dynamic)? I may not be
>following what you are saying; if so, let me know.

My own vote would be to set BLimit to at least four packets, regardless
of packet size.  Maybe even five packets...


>> * Maxburst:
>> "An additional drawback of MaxBurst ...".
>> What was the first drawback?  Introducing a separate control?  But
>> the earlier sentence says that "using two different controls may
>> make sense".
>
>New text:
>However, a drawback with
>MaxBurst is that adding a second control brings with it the
>possibility of the two transmission controllers interacting
>poorly to cause undesirable side effects.

I still don't buy it.  You have said elsewhere that 
"When BLimit == MaxBurst, CWL and MaxBurst perform similarly [AB05]."
Except that you think that MaxBurst might have a fatal flaw not
shared by CWL, you just haven't run across it yet?  Because you can
bring up the language of "two controllers interacting" for MaxBurst?
I don't buy it.  [Just as an aside, after the sentence above,
you could mention Aggressive MaxBurst, which improves on CWL by 
being robust to reordering.]

>> "the two transmission controllers may interact poorly, causing
>> undesirable side effects."
>> Do you have any examples of this?  Or citations?  Or anything
>> else to back this up?
>
>Not really. The idea was to suggest that there is the possibility of poor
>interaction due to the introduction of a new control. Maybe the new text
>(see above) captures this more effectively?

As I said above, I don't buy it...


...
>> Depending on how it is implemented, MaxBurst could be made to work
>> quite well with re-ordering, where each duplicate ack packet (possibly
>> from reordering of data packets) could be responded-to with up to
>> MaxBurst data packets (if allowed by the congestion window).
>> (E.g., Aggressive MaxBurst from [AB05].)
>> Can something like this be done with CWL as well?
>
>Hmm. Offhand, I do not see how this can be done with CWL, but I will think
>about it more. This is a good point though. Maybe we should mention this
>in the related work section where we discuss pros and cons of MaxBurst.

I think that would be good.

Regards,

- Sally

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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

This version includes all pending feedback based on Ted (provided and
addressed privately) and Pekka (posted recently).

I would like to try to get this wrapped up soon - perhaps the chairs can
suggest how to proceed.

Joe

Internet-Drafts@ietf.org wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
> This draft is a work item of the TCP Maintenance and Minor Extensions Working Group of the IETF.
> 
> 	Title		: Defending TCP Against Spoofing Attacks
> 	Author(s)	: J. Touch
> 	Filename	: draft-ietf-tcpm-tcp-antispoof-03.txt
> 	Pages		: 27
> 	Date		: 2006-2-22
> 	
> Recent analysis of potential attacks on core Internet infrastructure 
> indicates an increased vulnerability of TCP connections to spurious 
> resets (RSTs), sent with forged IP source addresses (spoofing).  TCP 
> has always been susceptible to such RST spoofing attacks, which were 
> indirectly protected by checking that the RST sequence number was 
> inside the current receive window, as well as via the obfuscation of 
> TCP endpoint and port numbers.  For pairs of well-known endpoints 
> often over predictable port pairs, such as BGP or between web servers 
> and well-known large-scale caches, increases in the path bandwidth-
> delay product of a connection have sufficiently increased the receive 
> window space that off-path third parties can guess a viable RST 
> sequence number.  The susceptibility to attack increases as the 
> square of the bandwidth, thus presents a significant vulnerability 
> for recent high-speed networks.  This document addresses this 
> vulnerability, discussing proposed solutions at the transport level 
> and their inherent challenges, as well as existing network level 
> solutions and the feasibility of their deployment.  This document 
> focuses on vulnerabilities due to spoofed TCP segments, and includes 
> a discussion of related ICMP spoofing attacks on TCP connections.
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-tcpm-tcp-antispoof-03.txt
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Sally Floyd wrote:
...
> My own vote would be to set BLimit to at least four packets, regardless
> of packet size.  Maybe even five packets...

The reason we chose that number is documented in our earlier ID. IMO,
much of that discussion is directly relevant to this work. In fact, I
can't see why this algorithm was derived from CWL vs BOL.

...
>> in the related work section where we discuss pros and cons of MaxBurst.

That too would be a good place to consider the discussion in the earlier ID.

IMO, it would be useful to merge those two documents to move forward,
since that doc had a much more detailed discussion of these alternatives.

Joe

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Folks,

I have submitted a revision of the tcp soft errors draft (now 
draft-ietf-tcpm-tcp-soft-errors-00.txt). The draft will be on the 
internet-drafts repository very soon. In the mean time, you can 
access it from: 
http://www.gont.com.ar/drafts/draft-ietf-tcpm-tcp-soft-errors-00.txt

I think the draft is pretty stable now. The latest feedback addresses 
was miscellaneous editorial changes proposed by Pekka Savola.

It would be very appreciated to receive feedback on this version.

The Abstract of the draft is:
---- cut here ----
    This document discusses problems that may arise due to TCP's reaction
    to soft errors.  In particular, it discusses the problem of long
    delays in connection establishment attempts that may arise when dual
    stack nodes that have IPv6 enabled by default are deployed in IPv4 or
    mixed IPv4 and IPv6 environments.  The purpose of this document is to
    discuss this potential problem, and analyze the ways in which it
    could be worked around.  It does not to try to specify whether IPv6
    should be enabled by default or not.
---- cut here ----


The course of the draft was basically this:

* draft-gont-tcpm-tcp-soft-errors-00.txt proposed the alternative 
processing of ICMP soft errors to avoid long delay between 
connection-establishment attempts. The WG agreed more discussion was 
needed on the pros and cons.

* draft-gont-tcpm-tcp-soft-errors-01.txt included alternative ways to 
avoid the long delays between connection establishment attempts 
(several connections issued in parallel, etc.), and a discussion of 
the pros and cons of the proposed mechanism. The WG agreed to take 
the document as a WG item, provided it just discussed the alternative 
processing of ICMP soft errors, rather than "proposing" it.

* draft-gont-tcpm-tcp-soft-errors-02.txt addressed those comments. 
This version of the draft does not propose the change, but rather 
limits itself to describe it and analyze its pros and cons.

* draft-ietf-tcpm-tcp-soft-errors-00.txt basically makes some 
miscellaneous editorial changes proposed by Pekka.

(All previous versions of the draft are available at: 
http://www.gont.com.ar/drafts/tcp-soft-errors.html)

Again, comments on the draft will be more than welcome.

Thanks!

--
Fernando Gont
e-mail: fernando@gont.com.ar || fgont@acm.org






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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 TCP Maintenance and Minor Extensions Working Group of the IETF.

	Title		: TCP's Reaction to Soft Errors

	Author(s)	: F. Gont
	Filename	: draft-ietf-tcpm-tcp-soft-errors-00.txt
	Pages		: 14
	Date		: 2006-2-24
	
This document discusses the problem of long delays between connection
establishment attempts that may arise in a number of scenarios,
including that in which dual stack nodes that have IPv6 enabled by
default are deployed in IPv4 or mixed IPv4 and IPv6 environments.
Additionally, it describes a modification to TCP's reaction to soft
errors that has been implemented in a variety of TCP/IP stacks to
help overcome this problem.


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Ron Bonica wrote:
> Folks,
>
> Please review the draft at the following URL:
>
>    http://www.ietf.org/internet-drafts/draft-bonica-tcp-auth-04.txt
>

Overall, this document still doesn't address the significant concerns
that Ted and I raised in on the -00 version, in particular, this seems
too much like a patch of TCP/MD5 without addressing the lack of a keying
mechanism.

In particular, it focuses too much (IMO) on time-based key rotation
rather than relying on secure, in-band rekeying.

Some particular comments follow:

key-chain is a TCP-EA data structure, not a TCP one

keys should NOT be used for both inbound and outbound simultaneously

what is the purpose of parsing the TCP header before validating the
segment? (the header has no meaning until validated)

why should routers log errors? they terminate TCP connections only when
acting as hosts anyway; it seems like all hosts should do this

TCP uses state and relative time (timeouts); this protocol uses absolute
time. the document AGAIN fails to discuss the security implications of
relying on absolute time (this was raised at the WG meeting in Vancouver)

	FWIW, RFC3562's reference to key lifetimes is NOT an endorsement
	of using absolute time to change keys; that doc is referring
	to rekeying over time (i.e., sec 16 is incorrect IMO -
	3562 never mentions 'schedules')

	I noted before that time-based keychains are not typical;
	it would be useful to explain why such a complex mechanism
	is being added, rather than a true rekeying protocol (which is
	more typical, and would address initial keying as well)

at least one key selection algorithm MUST be supported. if all MAYs are
dropped, the result MUST (IMO) be complete enough to work

regarding the MAC, are any fields padded if not word-aligned? (that
needs to be noted, either way)

when omitting the TCP options, does that also omit the TCP-EA option?
(it should not)

the algorithm does NOT need to be encoded in the header; the key ID can
easily include that information (like in IPsec)

the header could be much more compact:
	kind 	8 bits
	length	6 bits (40 is max value)
	K-bit	1 bit
	resv'd	1 bit	
		FYI - this MUST be set to zero in current impl's
		AND MUST be ignored on receipt
		("MUST be equal to zero" implies the
		receiver will check it, and that will kill
		its use for extensions)

	IMO, use all 16 bits remaining for the key ID
	(there are performance reasons to make the key ID
	space large and harder to guess off-path; I can discuss this
	at the WG meeting if anyone likes)

sec 11 should be calleed extensions, since this option is already named
an enahancement; however - this section MUST be completed for this doc
to have any impact.

'poor clock sync' is to colloquial; if you depend on clock sync, you need:
	- to discuss how you will establish that with only TCP
		or will this run only in the presence of NTP?
	- to discuss exactly how this depends on NTP

Errors in clock sync ARE errors detected here; that MAY be logged as a
TCP-EA error (otherwise you'll never be able to debug this)

the section on resets is incomplete; if you depend on time, you may
never be able to clean up state even if you have keys. that can affect
the ability to reuse port space. i.e., you may want to add something
that helps TCP-EA connections clean up if idle past when ANY key would
be valid (among other things)

the section on performance is too colloqial, esp. since perfomance is
one of the reasons TCP/MD5 is not widely used. it would help to have
current examples of the impact of performance on utility, even if they
change over time (FWIW, RFC1810 showed how they don't much)

end of sec 12.4 - the remaining octets are for other options (if there
are any others that fit in that space) or for larger hashes in the MACs
of TCP-EA.

sec 12.5 - if this is really intended to force exclusion of TCP/MD5, why
not use the same option number and use a flag to redefine the rest of
the header?

sec 12.6 - this defends against only off-path ICMP attacks; ICMP
messages can still be forged on-path with copies of valid packets and
not be invalidated.

sec 12.8 should be added - to describe relationship to IPsec/IKE

sec 13 changes the valid time of an existing key; that MUST NOT be
permitted, IMO. that sort of thing encourages key reuse, which defeats
the key lifetime issues addressed in RFC3562.

Along those lines, NO KEY should be able to have an expiration time that
is more than a few weeks in the future. in particular, "INFINITY" MUST
NOT be a valid time.



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From tcpm-bounces@ietf.org Mon Feb 27 05:42:41 2006
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Folks,

I have submitted a revision of the ICMP attacks draft. The draft will 
be on the internet-drafts repository very soon. In the mean time, you 
can access it from: 
http://www.gont.com.ar/drafts/draft-ietf-tcpm-icmp-attacks-00.txt

The Abstract of the draft is:
---- cut here ----
    This document discusses the use of the Internet Control Message
    Protocol (ICMP) to perform a variety of attacks against the
    Transmission Control Protocol (TCP) and other similar protocols.  It
    proposes several counter-measures to eliminate or minimize the impact
    of these attacks.
---- cut here ----

draft-gont-tcpm-icmp-attacks-05.txt was accepted as a WG document, 
for the Informational path. This (draft-tcpm-icmp-attacks-00.txt) new 
revision of the draft addresses the feedback received on at the last 
IETF and on the mailing-list. Among the changes are:
* RFC2119 wording was removed, to make the draft suitable for 
publication as an Informational RFC.
* The rationale behind each check has (hopefully) been clarified.
* Other checks that should be performed on the received ICMP messages 
(such as the checksum of the IP packet in the ICMP payload) are now 
included in the draft.

This draft has been around for quite a while, and has benefited from 
a large amount of feedback provided by both the TCPM WG, the open 
source community, vendors in general, etc. Most systems implement at 
least a subset of the counter-measures proposed in the draft.

Feedback on this revision of the draft will be more than welcome.

Thanks!

--
Fernando Gont
e-mail: fernando@gont.com.ar || fgont@acm.org






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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 TCP Maintenance and Minor Extensions Working Group of the IETF.

	Title		: ICMP attacks against TCP
	Author(s)	: F. Gont
	Filename	: draft-ietf-tcpm-icmp-attacks-00.txt
	Pages		: 36
	Date		: 2006-2-27
	
This document discusses the use of the Internet Control Message
Protocol (ICMP) to perform a variety of attacks against the
Transmission Control Protocol (TCP) and other similar protocols.  It
proposes several counter-measures to eliminate or minimize the impact
of these attacks.


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Hi.

Sorry to be so late with this, but if you're interested in presenting at
the TCPM meeting in Dallas, please let Mark <mallman@icir.org> and me
<faber@isi.edu> know before Thursday.  We'll post an agenda Friday.

Thanks!

--=20
Ted Faber
http://www.isi.edu/~faber           PGP: http://www.isi.edu/~faber/pubkeys.=
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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 TCP Maintenance and Minor Extensions Working Group of the IETF.

	Title		: A Roadmap for TCP Specification Documents
	Author(s)	: M. Duke, et al.
	Filename	: draft-ietf-tcpm-tcp-roadmap-06.txt
	Pages		: 42
	Date		: 2006-2-28
	
This document contains a "roadmap" to the Requests for Comments (RFC)
documents relating to the Internet's Transmission Control Protocol
(TCP).  This roadmap provides a brief summary of the documents
defining TCP and various TCP extensions that have accumulated in the
RFC series.  This serves as a guide and quick reference for both TCP
implementers and other parties who desire information contained in
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implementation to automatically retrieve the ASCII version of the
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ENCODING mime
FILE /internet-drafts/draft-ietf-tcpm-tcp-roadmap-06.txt

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

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