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Date: Wed, 3 Jan 2007 16:53:43 -0800
From: Stephen Hemminger <shemminger@osdl.org>
To: Ted Faber <faber@ISI.EDU>
Subject: Re: [tcpm] encouraging wider participation
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On Mon, 27 Nov 2006 13:27:24 -0800
Ted Faber <faber@ISI.EDU> wrote:

> On Mon, Nov 27, 2006 at 02:31:52AM -0300, Fernando Gont wrote:
> > At 20:56 22/11/2006, Duke, Martin wrote:
> > 
> > >If IETF would like to coordinate more with Linux, is the right move to
> > >send IETF reps to the next Linux meeting, to tell them what we're
> > >working on and how it might be useful to them?
> > 
> > I'd argue that, if anything, somebody from each major open source OS 
> > should be sent to the IETF meetings. That way, they could share their 
> > implementation/real-world experience in the meetings.
> 
> Make it so.
> 
> (See the problem?)
> 

Several core networking developers already go to IETF meetings.
Jamal Hadi Salam who worked on Traffic Control, and Yoshifuji from Linux IPV6
project attends as well.

-- 
Stephen Hemminger <shemminger@osdl.org>

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Date: Thu, 28 Dec 2006 14:23:03 +0100
From: Michael Scharf <michael.scharf@ikr.uni-stuttgart.de>
To: Lloyd Wood <L.Wood@surrey.ac.uk>
Subject: Re: [tcpm] Proposing ACKNOW bit in TCP header
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On Tue, 21 Nov 2006 at 15:41:04, Lloyd Wood wrote:
> >FWIW, Linux has had ABC for quite some time now as well.  First it was 
> >enabled by default, but no it's disabled now due to the performance 
> >problems in lots of applications which seem to rely on earlier 
> >behaviour.
> 
> Do you have a pointer to information on the problems with applications?
> 
> I'd been presuming that ABC is universally a good thing, even though its use blocks the count-the-acks-not-the-bytes sender hole that allows simple ack-splitting intercept-one-ack-send-two 'TCP acceleration', which isn't outlined in RFC3135, but is implemented in Cisco IOS:
> http://www.cisco.com/en/US/products/ps6441/products_feature_guide09186a0080683d9d.html

Performance problems might also occur with typical signaling
applications, e. g., in IP telephony platforms:

We have performed measurements on the end-to-end delay of small-sized
signaling messages for the case that a (small) percentage of messages
get lost, i. e., head-of-line blocking effects occur. We have studied
this effect both for TCP and SCTP with Linux kernel 2.6.15, and we
have compared the measurement results to the theoretical optimum.

We found that using ABC in Linux TCP results in significantly higher
latencies. Some of our findings can be found in the following
presentation from IEEE Globecom 2006:

  http://www.ikr.uni-stuttgart.de/~scharf/globecom2006_scharf_kiesel.pdf

Details on the theoretical model can be found in the corresponding paper:

  Michael Scharf, Sebastian Kiesel: "Head-of-Line Blocking in TCP and
  SCTP: Analysis and Measurements", Proc. IEEE Globecom, Nov. 2006

Note that TCP performs significantly worse than SCTP in our setup.

Best regards,

Michael Scharf and Sebastian Kiesel

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Subject: Re: [tcpm] spurious timeout reaction
From: Pasi Sarolahti <pasi.sarolahti@nokia.com>
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On Fri, 2006-12-15 at 15:00 -0500, ext Mark Allman wrote:
> We think the document is in decent shape and that it might make a good
> standards track mechanism for reacting to spurious timeouts.  The basic
> idea is that we say a TCP MAY make the RTO estimate more conservative in
> the face of spurious timeouts with the hopes of avoiding future
> timeouts.  In particular, we believe that since RFC2988 explicitly
> allows a TCP to be more conservative than the estimator given in that
> RFC that this reaction is already allowed and a stack using it would be
> considered compliant.

Wouldn't it be better to make this Experimental at first, like the other
related RFCs are?

I have to say I'm (still) somewhat unconvinced about this scheme. If one
link on the path is having delay spikes, it does not make the other
links any less likely to have loss patterns that cause legitimate
timeouts. Also, a wireless host that suffers from delay spikes can also
suffer from wireless packet losses that may cause legitimate timeouts
(that are not an indication of congestion) -- for example in a case
where a wireless host is able to use both Wireless LAN (RTTs of about
few milliseconds) and GPRS (RTTs in the range of 500 - 1000 ms +
occasional delay spikes) links, which is a fairly common case in the
recent mobile terminals. When a host is getting to the edge of the
wireless LAN coverage, it starts suffering from packet losses that could
easily cause a timeout before the host gets a new access interface up
and running. On the other hand, while using the GPRS interface the same
TCP connection can suffer from delay spikes.

So I don't think that delay spikes and legitimate timeouts during the
same TCP connection are necessarily as unlikely phenomenon as Section 4
on the disadvantages indicates (with a note that a legitimate RTO is not
necessarily a signal of bad congestion, like the text says -- it could
be an indication of loss of entire window on a WLAN link that just got
out of coverage, or loss of a retransmission).

I was also wondering about the dynamics of K' in the above case, because
hand-off to a high-latency link can also cause a spurious timeout.
Consider a host using WLAN for a while, so that the SRTT becomes around
5-10 ms and RTTVAR a few ms. Then the host moves to use GPRS where the
RTT is around 500-1000 ms and RTTVAR would maybe be in the ballpark of
100 ms. If the first RTT sample on the GPRS link is a longer one (as it
often is), the following scenario could be possible

K' =3D ceil ((R' - SRTT_prev) / RTTVAR_prev)
could make
ceil ((1000 - 10) / 3)  =3D  330

and after a while on the GPRS link the RTO would be around
RTO <- SRTT + max (G, K*RTTVAR), that could make (with an example SRTT
of 700 ms)
700 ms + max (G, 330 * 100 ms)  =3D  33700 ms

(did I get everything right above?)

So I believe there are possible real-world scenarios where this scheme
could kill the RTO performance, which is another reason why I would
prefer this becoming Experimental rather than Standards Track.

- Pasi


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From: Ted Faber <faber@ISI.EDU>
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On Wed, Jan 03, 2007 at 04:53:43PM -0800, Stephen Hemminger wrote:
> On Mon, 27 Nov 2006 13:27:24 -0800
> Ted Faber <faber@ISI.EDU> wrote:
> > On Mon, Nov 27, 2006 at 02:31:52AM -0300, Fernando Gont wrote:
> > > I'd argue that, if anything, somebody from each major open source OS=
=20
> > > should be sent to the IETF meetings. That way, they could share their=
=20
> > > implementation/real-world experience in the meetings.
> >=20
> > Make it so.
> >=20
> > (See the problem?)
>=20
> Several core networking developers already go to IETF meetings.
> Jamal Hadi Salam who worked on Traffic Control, and Yoshifuji from Linux =
IPV6
> project attends as well.

I believe that OS network implementors attend.  I intended to point out
that *compelling* that participation is problematic at best.  It's the
"should be sent" that's hard to make so. :-)

--=20
Ted Faber
http://www.isi.edu/~faber           PGP: http://www.isi.edu/~faber/pubkeys.=
asc
Unexpected attachment on this mail? See http://www.isi.edu/~faber/FAQ.html#=
SIG

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At 19:41 04/01/2007, Ted Faber wrote:

> > Several core networking developers already go to IETF meetings.
> > Jamal Hadi Salam who worked on Traffic Control, and Yoshifuji 
> from Linux IPV6
> > project attends as well.
>
>I believe that OS network implementors attend.  I intended to point out
>that *compelling* that participation is problematic at best.  It's the
>"should be sent" that's hard to make so. :-)

IMHO, the main reason for which many implementers don't come to (or 
actively participate in) the IETF is because participation usually 
leads into electro-political discussions.

I have seen many instances in which pretty straightforward stuff 
leads to endless discussions.
A current thread in the IETF list is one example. Discussions we have 
had here during the last couple of years is just another.
So it's really hard to come to the IETF, and at the same time be 
productive for the OS in which you work on.

And, when you read claims that "if you implement X, and it's not in 
the RFCs, it's a bug", you get even more reasons for doing the above.

Kindest regards,

--
Fernando Gont
e-mail: fernando@gont.com.ar || fgont@acm.org
PGP Fingerprint: 7809 84F5 322E 45C7 F1C9 3945 96EE A9EF D076 FFF1






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well said Fernando! analysis paralysis is the name of the
game!
=20
> -----Original Message-----
> From: Fernando Gont [mailto:fernando@gont.com.ar]=20
> Sent: Thursday, January 04, 2007 6:31 PM
> To: Ted Faber; Stephen Hemminger
> Cc: tcpm@ietf.org
> Subject: Re: [tcpm] encouraging wider participation
>=20
> At 19:41 04/01/2007, Ted Faber wrote:
>=20
> > > Several core networking developers already go to IETF meetings.
> > > Jamal Hadi Salam who worked on Traffic Control, and Yoshifuji
> > from Linux IPV6
> > > project attends as well.
> >
> >I believe that OS network implementors attend.  I intended=20
> to point out=20
> >that *compelling* that participation is problematic at best.=20
>  It's the=20
> >"should be sent" that's hard to make so. :-)
>=20
> IMHO, the main reason for which many implementers don't come=20
> to (or actively participate in) the IETF is because=20
> participation usually leads into electro-political discussions.
>=20
> I have seen many instances in which pretty straightforward=20
> stuff leads to endless discussions.
> A current thread in the IETF list is one example. Discussions=20
> we have had here during the last couple of years is just another.
> So it's really hard to come to the IETF, and at the same time=20
> be productive for the OS in which you work on.
>=20
> And, when you read claims that "if you implement X, and it's=20
> not in the RFCs, it's a bug", you get even more reasons for=20
> doing the above.
>=20
> Kindest regards,
>=20
> --
> Fernando Gont
> e-mail: fernando@gont.com.ar || fgont@acm.org PGP=20
> Fingerprint: 7809 84F5 322E 45C7 F1C9 3945 96EE A9EF D076 FFF1
>=20
>=20
>=20
>=20
>=20
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www1.ietf.org/mailman/listinfo/tcpm
>=20

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To: Pasi Sarolahti <pasi.sarolahti@nokia.com>
From: Mark Allman <mallman@icir.org>
Subject: Re: [tcpm] spurious timeout reaction 
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Pasi-

Thanks for the note.  Some thoughts ...

> On Fri, 2006-12-15 at 15:00 -0500, ext Mark Allman wrote:
> > We think the document is in decent shape and that it might make a
> > good standards track mechanism for reacting to spurious timeouts.
> > The basic idea is that we say a TCP MAY make the RTO estimate more
> > conservative in the face of spurious timeouts with the hopes of
> > avoiding future timeouts.  In particular, we believe that since
> > RFC2988 explicitly allows a TCP to be more conservative than the
> > estimator given in that RFC that this reaction is already allowed
> > and a stack using it would be considered compliant.
> 
> Wouldn't it be better to make this Experimental at first, like the
> other related RFCs are?

IMO, no.

I have a few reasons for that opinion:

(1) As noted above, nothing we suggest in this document would make a TCP
    non-conformant with RFC 2988.  So, as far as I am concerned RFC or
    not this change could be implemented by a stack today and nobody
    could complain about it.

(2) This change would not be a MUST.  That is, implementers could use
    this scheme or not.

(3) I think we are way too conservative with the "PS" designation.  We
    seem to have this feeling that everything needs a run at
    experimental first without any real reason.  Some things certainly
    should take a spin there.  And, where the line should be is clearly
    fuzzy.  I would personally draw it differently than the community
    has drawn it.

(4) The change is not overall hurtful to the network.

> If one link on the path is having delay spikes, it does not make the
> other links any less likely to have loss patterns that cause
> legitimate timeouts. Also, a wireless host that suffers from delay
> spikes can also suffer from wireless packet losses that may cause
> legitimate timeouts (that are not an indication of congestion) -- for
> example in a case where a wireless host is able to use both Wireless
> LAN (RTTs of about few milliseconds) and GPRS (RTTs in the range of
> 500 - 1000 ms + occasional delay spikes) links, which is a fairly
> common case in the recent mobile terminals. When a host is getting to
> the edge of the wireless LAN coverage, it starts suffering from packet
> losses that could easily cause a timeout before the host gets a new
> access interface up and running. On the other hand, while using the
> GPRS interface the same TCP connection can suffer from delay spikes.

I understand that we can think about scenarios where anything can cause
problems.  And, it is not my goal to cause problems here, of course.
But, I really have to wonder about all this and how much of it is *real*
and how much of it is just *possible*.  (And, even for things that are
real one also wonders about the prevalence.  Seemingly every corner case
happens sometimes or somewhere, but I am not sure that should drive our
engineering.) 

We explicitly added a bit to the document that says one could not use K'
when the connection would be for sure in the regime where it would be
using the RTO for all loss recovery.  I guess it is not clear to me what
these other scenarios are that really dictate the use of the RTO.  We
know how to do pretty robust loss recovery without relying on the RTO.

And, recall that by increasing the K we are avoiding spurious timeouts
that ultimately collapse the window, cause some go-back-N and generally
do not help connection performance at all.

> So I don't think that delay spikes and legitimate timeouts during the
> same TCP connection are necessarily as unlikely phenomenon as Section
> 4 on the disadvantages indicates (with a note that a legitimate RTO is
> not necessarily a signal of bad congestion, like the text says -- it
> could be an indication of loss of entire window on a WLAN link that
> just got out of coverage, or loss of a retransmission).

Losing an entire window is bad congestion.

I understand that is not what you meant to convey.  But, the thing is
.... loss == congestion is the assumption built into TCP.  So, when the
assumption is busted things break.  I am not sure that our particular
technique can or should be held up because the underlying assumption
sometimes does not hold.

(Also note that I am not saying it is ideal that we assume loss ==
congestion.  Clearly it is not.  And, clearly work on changing this is
useful.  I am just saying that is the current situation and that I do
not buy 'the assumption is busted' as a really great argument against
what we said in our document.)

> I was also wondering about the dynamics of K' in the above case,
> because hand-off to a high-latency link can also cause a spurious
> timeout.  Consider a host using WLAN for a while, so that the SRTT
> becomes around 5-10 ms and RTTVAR a few ms. Then the host moves to use
> GPRS where the RTT is around 500-1000 ms and RTTVAR would maybe be in
> the ballpark of 100 ms. If the first RTT sample on the GPRS link is a
> longer one (as it often is), the following scenario could be possible
> 
> K' = ceil ((R' - SRTT_prev) / RTTVAR_prev)
> could make
> ceil ((1000 - 10) / 3)  =  330
> 
> and after a while on the GPRS link the RTO would be around
> RTO <- SRTT + max (G, K*RTTVAR), that could make (with an example SRTT
> of 700 ms)
> 700 ms + max (G, 330 * 100 ms)  =  33700 ms
> 
> (did I get everything right above?)

Before coffee this looks mostly right.  If one thinks the RTO min is
being used then the numbers above might not cause a spurious RTO.  But,
if one increased the spike a little then I think the argument makes
sense. 

One point is that if you ever dropped back to the WLAN you'd be
protected.  

(And, per the above, I am not sure I completely buy this case ...)

A couple of thoughts ...

  + The draft could say that a way to decide to use K' would be to first
    assess the number of RTOs that would be impacted.  So, e.g., if a
    TCP connection never RTOed and then had a spurious RTO because of a
    delay spike then it might make fine sense to use the K'.  On the
    other hand, if the spurious RTO was one 1 out of 100 RTOs and the K'
    ended up big (as above) then the TCP might decide to just not use
    the K'.  The problem is how to do the fuzzy middle of this space.

    I am not saying that we'd have to work out some scheme to do this.
    I am just wondering if, since this is not presented to be mandatory,
    whether we can add a little bit of explicit wiggle room for
    implementers.

  + We could keep the fundamental notion that the RTO estimation has
    failed, but change the particulars of how we cope.  

    Right now we make the argument that delay spikes that cause spurious
    RTOs indicate that the variance estimator is not coping with the
    variance in the RTT all that well.  I think that is a reasonable
    line of thinking.  But, what if we decided that the variance
    component was not supposed to catch delay spikes and added another
    term to catch these things if we experienced any that caused
    spurious timeouts?  So, if we have an R' from a spurious timeout and
    an RTO_prev, we could do something like this:

        DS = R' - RTO
        RTO = SRTT + max (G, K*RTTVAR) + DS
            (which is the formula from RFC2988 with "+ DS" added)

    Would that give us the same amount of protection without giving a
    multiplicative effect that balloons the RTO to ridiculousness?

  + Also, there are schemes floating around that would essentially
    re-init all the CC parameters on network change.  It seems that it
    might be useful to include any K' in that re-initialization such
    that the above case does not happen.  That is, when we can explain
    the delay spike and take a different approach to ensure that
    spurious RTOs are not repeated then we don't have to kludge up the
    RTO to try to cope.  It is only when we cannot explain the delay
    spike that we try to cope using the RTO.  Make any sense?

Further thoughts?

Thanks,
allman




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Hi,

I just re-read the draft more carefully, and I like the approach. I've
been thinking that this technique of adjusting K' might be enough to
eliminate the special case step (2.4) in RFC 2988.

I've always disliked the min RTO = 1 sec rule. I understand why it is
there, but the common case on the net seems to usually hit this step. In
other words, we do all this work to calculate the RTO to finally find
that we have to just set RTO = 1 sec.

Any thoughts?

-- 
Armando
www.armandocaro.net


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Subject: Re: [tcpm] spurious timeout reaction
From: Pasi Sarolahti <pasi.sarolahti@nokia.com>
To: mallman@icir.org
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(I'm chopping off the original mail here and there)

On Mon, 2007-01-08 at 10:28 -0500, ext Mark Allman wrote:=20
> > Wouldn't it be better to make this Experimental at first, like the
> > other related RFCs are?
>=20
> IMO, no.
>=20
> I have a few reasons for that opinion:
>=20
> (1) As noted above, nothing we suggest in this document would make a TCP
>     non-conformant with RFC 2988.  So, as far as I am concerned RFC or
>     not this change could be implemented by a stack today and nobody
>     could complain about it.
>=20
> (2) This change would not be a MUST.  That is, implementers could use
>     this scheme or not.
>
> (3) I think we are way too conservative with the "PS" designation.  We
>     seem to have this feeling that everything needs a run at
>     experimental first without any real reason.  Some things certainly
>     should take a spin there.  And, where the line should be is clearly
>     fuzzy.  I would personally draw it differently than the community
>     has drawn it.

Yep, you can say that again... And sometimes it takes time to get even
Experimentals through. Speaking of PS, shouldn't the normative
references in a PS document also be standards track documents? That
would seem logical, at least. In particular, we are missing a spurious
timeout detection PS.

[...]
> I understand that we can think about scenarios where anything can cause
> problems.  And, it is not my goal to cause problems here, of course.
> But, I really have to wonder about all this and how much of it is *real*
> and how much of it is just *possible*.  (And, even for things that are
> real one also wonders about the prevalence.  Seemingly every corner case
> happens sometimes or somewhere, but I am not sure that should drive our
> engineering.)=20

A corner case for you might not be a corner case for me. The WLAN-GPRS
example is an existing case from real world, so for some people it is
mote than just a corner case. It is quite hard to determine what are the
globally typical path characteristics nowadays.

> We explicitly added a bit to the document that says one could not use K'
> when the connection would be for sure in the regime where it would be
> using the RTO for all loss recovery.  I guess it is not clear to me what
> these other scenarios are that really dictate the use of the RTO.  We
> know how to do pretty robust loss recovery without relying on the RTO.

Because TCP sender is able to detect spurious RTO, it also knows how
often genuine RTOs occur, i.e., how typical it is that the normal loss
recovery does not work as expected. One possibility could be that when
TCP sender detects a genuine RTO, it would reset K' back to the original
value.

> Losing an entire window is bad congestion.
>=20
> I understand that is not what you meant to convey.  But, the thing is
> .... loss =3D=3D congestion is the assumption built into TCP.  So, when t=
he
> assumption is busted things break.  I am not sure that our particular
> technique can or should be held up because the underlying assumption
> sometimes does not hold.

Yep. But even if we assume that genuine RTO is an indication of bad
congestion, I'm not sure if we can justify K' as an additional
congestion control mechanism. Retransmission timer already has the
backoff mechanism that is quite conservative. If the initial RTO
estimate was several seconds, with the normal RTO backoff and some bad
luck the TCP sender would need to wait for its turn for quite a while.
If your neighbor didn't happen to have spurious timeouts, you are in
quite bad position in terms of fairness.

> A couple of thoughts ...
>=20
>   + The draft could say that a way to decide to use K' would be to first
>     assess the number of RTOs that would be impacted.  So, e.g., if a
>     TCP connection never RTOed and then had a spurious RTO because of a
>     delay spike then it might make fine sense to use the K'.  On the
>     other hand, if the spurious RTO was one 1 out of 100 RTOs and the K'
>     ended up big (as above) then the TCP might decide to just not use
>     the K'.  The problem is how to do the fuzzy middle of this space.

The problem with the statistical methods is that they don't help the
short connections. And often the short connections are those where the
user can sense and get irritated by the relative effect of an RTO more
than in the long transfers.

>   + We could keep the fundamental notion that the RTO estimation has
>     failed, but change the particulars of how we cope. =20
>=20
>     Right now we make the argument that delay spikes that cause spurious
>     RTOs indicate that the variance estimator is not coping with the
>     variance in the RTT all that well.  I think that is a reasonable
>     line of thinking.  But, what if we decided that the variance
>     component was not supposed to catch delay spikes and added another
>     term to catch these things if we experienced any that caused
>     spurious timeouts?  So, if we have an R' from a spurious timeout and
>     an RTO_prev, we could do something like this:
>=20
>         DS =3D R' - RTO
>         RTO =3D SRTT + max (G, K*RTTVAR) + DS
>             (which is the formula from RFC2988 with "+ DS" added)
>=20
>     Would that give us the same amount of protection without giving a
>     multiplicative effect that balloons the RTO to ridiculousness?

This seems better to me. I think it is appropriate to treat a spurious
timeout as an exceptional event, rather than a component of the normal
RTT variance.

>   + Also, there are schemes floating around that would essentially
>     re-init all the CC parameters on network change.  It seems that it
>     might be useful to include any K' in that re-initialization such
>     that the above case does not happen.  That is, when we can explain
>     the delay spike and take a different approach to ensure that
>     spurious RTOs are not repeated then we don't have to kludge up the
>     RTO to try to cope.  It is only when we cannot explain the delay
>     spike that we try to cope using the RTO.  Make any sense?

Yes, it does. And it could be useful to refer to, for example, the RLCI
draft here, maybe with a note that since spurious timeouts have been
identified to be a characteristic of some wireless links, it could be
worthwhile to consider implementing also RLCI, if this is implemented.

- Pasi


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To: Pasi Sarolahti <pasi.sarolahti@nokia.com>
From: Mark Allman <mallman@icir.org>
Subject: Re: [tcpm] spurious timeout reaction 
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> Speaking of PS, shouldn't the normative
> references in a PS document also be standards track documents? 

Yes.

> That would seem logical, at least. In particular, we are missing a
> spurious timeout detection PS.

Yes.

> A corner case for you might not be a corner case for me. The WLAN-GPRS
> example is an existing case from real world, so for some people it is
> mote than just a corner case. It is quite hard to determine what are
> the globally typical path characteristics nowadays.

Of course you are right.

So, then how do we engineer things for a network that includes as much
heterogeneity as we currently have?

> > We explicitly added a bit to the document that says one could not
> > use K' when the connection would be for sure in the regime where it
> > would be using the RTO for all loss recovery.  I guess it is not
> > clear to me what these other scenarios are that really dictate the
> > use of the RTO.  We know how to do pretty robust loss recovery
> > without relying on the RTO.
> 
> Because TCP sender is able to detect spurious RTO, it also knows how
> often genuine RTOs occur, i.e., how typical it is that the normal loss
> recovery does not work as expected. One possibility could be that when
> TCP sender detects a genuine RTO, it would reset K' back to the
> original value.

I don't like this idea or really any idea that collapses K' back down.
The issue is that these delay spikes are *unforeseen*.  If we could
somehow predict that the RTO estimate was going to be too low then we'd
clearly do something about it or change the RTO calculation to
accommodate the fact that we could see a delay spike coming.  But, in
the absence of such information all we can do is look to our history
about the magnitude of delay spikes.  So, after collecting the history,
throwing it away seems like a bad idea to me.

> > Losing an entire window is bad congestion.
> > 
> > I understand that is not what you meant to convey.  But, the thing is
> > .... loss == congestion is the assumption built into TCP.  So, when the
> > assumption is busted things break.  I am not sure that our particular
> > technique can or should be held up because the underlying assumption
> > sometimes does not hold.
> 
> Yep. But even if we assume that genuine RTO is an indication of bad
> congestion, I'm not sure if we can justify K' as an additional
> congestion control mechanism. Retransmission timer already has the
> backoff mechanism that is quite conservative. If the initial RTO
> estimate was several seconds, with the normal RTO backoff and some bad
> luck the TCP sender would need to wait for its turn for quite a while.
> If your neighbor didn't happen to have spurious timeouts, you are in
> quite bad position in terms of fairness.

I am not quite sure I follow this.  K' is not a congestion control
mechanism.  The RTO is the congestion control mechanism.  If you have an
increased K (over the standard 4) and are still losing packets and have
to backoff the RTO further, you're going to have a hard time convincing
me that the increased K is really a big problem.  I.e., we're in a bad
situation and clearly K' is not a factor.

The point of K' is to try to avoid needless timeouts, not to in some
what shape a connection's rate or fair hare or whatever.  Certainly it
may take longer for legit RTOs.  But, a connection also may prevent the
ill-effects of a needless RTO.

> > A couple of thoughts ...
> > 
> >   + The draft could say that a way to decide to use K' would be to first
> >     assess the number of RTOs that would be impacted.  So, e.g., if a
> >     TCP connection never RTOed and then had a spurious RTO because of a
> >     delay spike then it might make fine sense to use the K'.  On the
> >     other hand, if the spurious RTO was one 1 out of 100 RTOs and the K'
> >     ended up big (as above) then the TCP might decide to just not use
> >     the K'.  The problem is how to do the fuzzy middle of this space.
> 
> The problem with the statistical methods is that they don't help the
> short connections. And often the short connections are those where the
> user can sense and get irritated by the relative effect of an RTO more
> than in the long transfers.

Yep.  This technique is clearly for longer-term transfers.

(On the other hand, short connections might also always be in the regime
the document declares as off-limits for a K'.  I.e., a small cwnd.)

Thanks!

allman




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In the ICCRG's goal to foster a long-term congestion control
architecture for the Internet, its initial step is to understand all of
the current mechanisms in use.  As an aid to this, the ICCRG is working
on a document that summarizes all of the congestion control mechanisms
and other guidance that the IETF has produced and published in the RFC
series.  The initial version of this document is online:
http://tools.ietf.org/group/irtf/draft-irtf-iccrg-cc-rfcs-00.txt

We think this is a fairly complete listing of the RFCs that relate to
congestion control, but if you know of others that should be included,
we would like to hear about it.

Also, comments on what can be done to make this more useful or readable
are welcome.  Comments on and improvements to the text summarizing each
RFC are also welcome.

The goal is to gather these inputs before the ICCRG meeting in
mid-February.  After the ICCRG meeting, the document will be updated
based on these comments and discussion from the meeting.

Please send all comments to the ICCRG list (iccrg@cs.ucl.ak.uk).  Although
this request is being sent to multiple lists, this is only to notify
people not on the ICCRG list; we do not want to burden the other mailing
lists with replies, so please trim the "to" and "cc" fields of replies.

Thanks in advance.

-- 
Wesley M. Eddy
Verizon Federal Network Systems

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Subject: [tcpm] DoS attack from misbehaving receivers
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Has anyone in this group explored the problems described in:
"Misbehaving TCP Receivers Can Cause Internet-Wide Congestion Collapse"?

	http://www.cs.umd.edu/~capveg/optack/optack-ccs05.pdf
	http://www.cs.umd.edu/~capveg/
	http://www.kb.cert.org/vuls/id/102014

A possible solution (optack) is described in the paper that involves the
sender randomly skipping segments and resetting connections that
ACK data that was never sent.  But it is not clear that the impacts
of such a change have been fully investigated.

Comments?

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Stephen Hemminger wrote:
> Has anyone in this group explored the problems described in:
> "Misbehaving TCP Receivers Can Cause Internet-Wide Congestion Collapse"=
?
>=20
> 	http://www.cs.umd.edu/~capveg/optack/optack-ccs05.pdf
> 	http://www.cs.umd.edu/~capveg/
> 	http://www.kb.cert.org/vuls/id/102014
>=20
> A possible solution (optack) is described in the paper that involves th=
e
> sender randomly skipping segments and resetting connections that
> ACK data that was never sent.  But it is not clear that the impacts
> of such a change have been fully investigated.

There's very little in TCP that is intended to work well with a
byzantine receiver. Except for our security protocols, very little of
the Internet deals well with them either.

Joe

--=20
----------------------------------------
Joe Touch
Sr. Network Engineer, USAF TSAT Space Segment


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From tcpm-bounces@ietf.org Thu Jan 11 15:05:37 2007
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Stephen Hemminger wrote:
> Has anyone in this group explored the problems described in:
> "Misbehaving TCP Receivers Can Cause Internet-Wide Congestion
> Collapse"?=20
>=20
> 	http://www.cs.umd.edu/~capveg/optack/optack-ccs05.pdf
> 	http://www.cs.umd.edu/~capveg/
> 	http://www.kb.cert.org/vuls/id/102014
>=20
> A possible solution (optack) is described in the paper that
> involves the sender randomly skipping segments and resetting
> connections that ACK data that was never sent.  But it is not
> clear that the impacts of such a change have been fully investigated.
>=20
> Comments?
>=20

The proposed test for a non-compliant receiver would seem to
require that the sender be non-compliant itself.

For many of the applications cited, wouldn't a more general
solution be to apply connection-specific rate limiters at
the sender (in the stack or at the application layer)?


Even if one client could truly receive at full line rate,
is is truly wise at the application layer to agree to give
all of your capacity to one client when hundreds or thousands
of clients are anticipated?



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From tcpm-bounces@ietf.org Thu Jan 11 16:00:06 2007
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Rob Sherwood wrote:
> I guess I should probably jump in here somewhere :-)
>=20
> On Thu, Jan 11, 2007 at 12:00:59PM -0800, Caitlin Bestler wrote:
>> The proposed test for a non-compliant receiver would seem to require
>> that the sender be non-compliant itself.
>=20
> I'm not sure I understand this comment.  Part of the utility
> of the solution is that unmodified receivers can communicate
> with senders that randomly skip segments, because segment
> reordering and dropping is already handled by TCP.
>=20

Having the sender skip segments will indeed detect a non-compliant
receiver that acks segments that were never sent. But sending
non-contiguous TCP segments is itself not compliant.

>=20
>> For many of the applications cited, wouldn't a more general solution
>> be to apply connection-specific rate limiters at the sender (in the
>> stack or at the application layer)?
>=20
> As mentioned in the paper, rate limiting does not prevent the attack.
> The attack is for many servers to send a small amount of TCP
> unfriendly traffic into the network, which results in
> significant traffic in aggregate.  Reducing the rate of
> traffic at any one server only causes the attacker to target
> more servers.
>=20

You can generate a large amount of traffic simply sending small
requests that generate large responses and then bit-bucketing
the results. If you are doing so from a botnet you can even
actually receive all of the segments in the distributed clients
before dropping the segments.

Before endorsing a counter-measure that calls for what was
previously non-compliant behavior there should be a showing
that the problem being solved is indeed severe and unique.

If my understanding is correct, the problem is that an attacker
can simulate reception of more material than their actual local
link can support, and thereby trick a large group of senders
into saturating *their* local links and/or intermediate segments.

But is it not correct that the same attack could be launched
from a botnet just as effectively even without faking acks?
Until typical home computers are more secure from being drafted
into a botnet is there a real benefit from this counter-measure?

Having transmitters engage in non-compliant behavour in order
to catch cheaters is a slippery slope best avoided.


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From tcpm-bounces@ietf.org Thu Jan 11 16:27:57 2007
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From: Rob Sherwood <capveg@cs.umd.edu>
To: Caitlin Bestler <caitlinb@broadcom.com>
Subject: Re: [tcpm] DoS attack from misbehaving receivers
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> Before endorsing a counter-measure that calls for what was
> previously non-compliant behavior there should be a showing
> that the problem being solved is indeed severe and unique.

Agreed.  This is the crux of the discussion: is this attack severe
enough to motivate the admittedly invasive change to the TCP stack.

> If my understanding is correct, the problem is that an attacker
> can simulate reception of more material than their actual local
> link can support, and thereby trick a large group of senders
> into saturating *their* local links and/or intermediate segments.

This is mostly correct.  The true concern is that *backbone* links,
not just local links, will be saturated, bringing about, as the title
suggests, Internet-wide congestion collapse.  It is this concern that
I feel motivates the need to change the specification.

> But is it not correct that the same attack could be launched
> from a botnet just as effectively even without faking acks?
> Until typical home computers are more secure from being drafted
> into a botnet is there a real benefit from this counter-measure?

A botnet of sufficient size, maybe.  An open question (read: subject of
my current research) is how much traffic is required to overcome Internet
backbone links.  The reason for concern with the OptAck attack is that
the amplification factors are large (~1600x and higher from the paper),
which reduces the size of the botnet required to cause significant damage.

For example, if a 100,000 node botnet has sufficient bandwidth to
overwhelm backbone links without OptAck, then a 60 node botnet can
overwhelm the network with OptAck.  I believe that there is a qualitative
difference between these two attacks in terms of attacker capability,
i.e., it takes significantly more work to compromise 100K machines
then 60.

> Having transmitters engage in non-compliant behavour in order
> to catch cheaters is a slippery slope best avoided.

In general I agree, but I believe that the danger here is significant
to warrant futher consideration.

- Rob
.


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Caitlin Bestler wrote:
> Rob Sherwood wrote:
>> I guess I should probably jump in here somewhere :-)
>>
>> On Thu, Jan 11, 2007 at 12:00:59PM -0800, Caitlin Bestler wrote:
>>> The proposed test for a non-compliant receiver would seem to require
>>> that the sender be non-compliant itself.
>> I'm not sure I understand this comment.  Part of the utility
>> of the solution is that unmodified receivers can communicate
>> with senders that randomly skip segments, because segment
>> reordering and dropping is already handled by TCP.
>=20
> Having the sender skip segments will indeed detect a non-compliant
> receiver that acks segments that were never sent. But sending
> non-contiguous TCP segments is itself not compliant.

Technically, that's equivalent to losing or reordering segments. As long
as you can't tell the difference from the receiver's point of view, why
is this not compliant?

Joe
--=20
----------------------------------------
Joe Touch
Sr. Network Engineer, USAF TSAT Space Segment


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Joe Touch wrote:
> Caitlin Bestler wrote:
>> Rob Sherwood wrote:
>>> I guess I should probably jump in here somewhere :-)
>>>=20
>>> On Thu, Jan 11, 2007 at 12:00:59PM -0800, Caitlin Bestler wrote:
>>>> The proposed test for a non-compliant receiver would seem to
>>>> require that the sender be non-compliant itself.
>>> I'm not sure I understand this comment.  Part of the utility of the
>>> solution is that unmodified receivers can communicate with senders
>>> that randomly skip segments, because segment reordering and dropping
>>> is already handled by TCP.
>>=20
>> Having the sender skip segments will indeed detect a non-compliant
>> receiver that acks segments that were never sent. But sending
>> non-contiguous TCP segments is itself not compliant.
>=20
> Technically, that's equivalent to losing or reordering
> segments. As long as you can't tell the difference from the
> receiver's point of view, why is this not compliant?
>=20
> Joe

There's a difference between L3 or L2 dropping a segment
because a queue was full and L4 simply not generating it.

Granted, there is no method to prove the difference when
observing from the other end of the network, but that still
isn't a reason to bless generation of non-compliant TCP streams.

A TCP connection is supposed to be a peer-to-peer relationship.
Once you start playing liar's poker to try to trip up the other
end things are going to get messy very quickly. What if receivers
started doing things to try to detect sender's who were sending
out of order?

Be paranoid in what you accept and devious in what you send?



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> > But is it not correct that the same attack could be launched
> > from a botnet just as effectively even without faking acks?
> > Until typical home computers are more secure from being drafted
> > into a botnet is there a real benefit from this counter-measure?
>=20
> A botnet of sufficient size, maybe.  An open question (read: subject
of
> my current research) is how much traffic is required to overcome
> Internet
> backbone links.  The reason for concern with the OptAck attack is that
> the amplification factors are large (~1600x and higher from the
paper),
> which reduces the size of the botnet required to cause significant
> damage.

I have witnessed distributed DOS attacks generating several Gbps of
traffic, and that was a couple of years ago. I have also witnessed DOS
attacks implemented by simply opening multiple TCP connections. The
attack software started several threads, and in each thread a loop would
keep loading a particular web page. That actually gives a lot of
amplification, since the size of the HTTP request is only a fraction of
the size of the response. There was no attempt to hide the origin of the
attack, spoof the IP address, hack the TCP stack, or any of that. These
particular attacks were trying to bring down a web site, but similar
attacks could easily target a particular link in the infrastructure.

-- Christian Huitema

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Caitlin Bestler wrote:
=2E..
>>> Having the sender skip segments will indeed detect a non-compliant
>>> receiver that acks segments that were never sent. But sending
>>> non-contiguous TCP segments is itself not compliant.
>> Technically, that's equivalent to losing or reordering
>> segments. As long as you can't tell the difference from the
>> receiver's point of view, why is this not compliant?
>>
>> Joe
>=20
> There's a difference between L3 or L2 dropping a segment
> because a queue was full and L4 simply not generating it.

Not to the receiver!

('goes to intent' - the receiver doesn't know intent).

> Granted, there is no method to prove the difference when
> observing from the other end of the network, but that still
> isn't a reason to bless generation of non-compliant TCP streams.

It's not non-compliant to do something that the receiver can't
differentiate from legitimate behavior.

> A TCP connection is supposed to be a peer-to-peer relationship.
> Once you start playing liar's poker to try to trip up the other
> end things are going to get messy very quickly. What if receivers
> started doing things to try to detect sender's who were sending
> out of order?

What would you do? You can't punish them, since you can't know whether
the network is the cause or not.

The better quote might be a version of the (in)famous "don't attribute
to malice that which can be attributed to ignorance"; my version here
would be:

"don't attribute to malice that which can be attributed to entropy"

Joe

--=20
----------------------------------------
Joe Touch
Sr. Network Engineer, USAF TSAT Space Segment


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Rob Sherwood wrote:
>> Before endorsing a counter-measure that calls for what was
>> previously non-compliant behavior there should be a showing
>> that the problem being solved is indeed severe and unique.
> 
> Agreed.  This is the crux of the discussion: is this attack severe
> enough to motivate the admittedly invasive change to the TCP stack.

Yep.


>> If my understanding is correct, the problem is that an attacker
>> can simulate reception of more material than their actual local
>> link can support, and thereby trick a large group of senders
>> into saturating *their* local links and/or intermediate segments.
> 
> This is mostly correct.  The true concern is that *backbone* links,
> not just local links, will be saturated, bringing about, as the title
> suggests, Internet-wide congestion collapse.  It is this concern that
> I feel motivates the need to change the specification.

The links with heavily aggregated traffic from local or regional ISPs up 
to the tier 1 ISPs are almost always bandwidth capped, and are almost 
always purchased in such that they are able to carry "just enough" 
traffic -- that is, it's pegged for at least some of the time.  (Who 
wants to pay for more than they use?)  The core of the Internet today is 
provisioned to handle the load.  It seems to me the threat of 
Internet-wide congestion collapse is a bit overstated these days.  (This 
also applies to recent congestion control discussions.)

I could see this attack as being effective at saturating something like 
a university's access link where a small number of well-connected 
servers could severely congest it.  However, in my experience, server 
and network admins are very reluctant to run any sort of service that 
will stream arbitrary amounts of data as fast as they can.  (I've 
sometimes tried to get such services made available for measurement 
purposes.)

The defense proposed in the paper is pretty cool, but implementing it 
and running it on a production machine would make me a bit nervous.  If 
this were a significant problem, you could always turn on bandwidth caps 
in the server application or in the kernel's networking system.  This 
would probably be my favored approach.  In reality, I've never heard of 
this attack being perpetrated in the wild, while DDOS is quite common.

   -John

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On Thu, Jan 11, 2007 at 05:15:24PM -0500, John Heffner wrote:
> The links with heavily aggregated traffic from local or regional ISPs up 
> to the tier 1 ISPs are almost always bandwidth capped, and are almost 
> always purchased in such that they are able to carry "just enough" 
> traffic -- that is, it's pegged for at least some of the time.  (Who 
> wants to pay for more than they use?)  The core of the Internet today is 
> provisioned to handle the load.  It seems to me the threat of 
> Internet-wide congestion collapse is a bit overstated these days.  (This 
> also applies to recent congestion control discussions.)

This is the part that is not clear to me: is it really the case that
back bone links have sufficient capacity to handle the sum of all
the access link capacity?  That is to say not the expected load, but
the maximum capacity?   I am familiar with many of the link capacity
estimation tools available, and it's my understanding that there is not
a conclusive answer here.

- Rob
.

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From: Gavin McCullagh <gavin.mccullagh@nuim.ie>
Subject: Re: [tcpm] DoS attack from misbehaving receivers
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On Thu, 11 Jan 2007, Rob Sherwood wrote:

> > Before endorsing a counter-measure that calls for what was
> > previously non-compliant behavior there should be a showing
> > that the problem being solved is indeed severe and unique.
> 
> Agreed.  This is the crux of the discussion: is this attack severe
> enough to motivate the admittedly invasive change to the TCP stack.

We implemented a basic attack of this recently before realising that others
had been here first.  I've thrown up a few quick results here using a
recent linux kernel on sender and receiver:

	http://www.hamilton.ie/gavinmc/drop_dupack_attack/

Kernel: 2.6.18 on both
OS: Debian GNU/Linux "edgy"
Apache v2.0.55
wget v1.10.2

The patch to linux is a short one.  I can make it available, but have held
off doing so for now.

Gavin


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From: "Caitlin Bestler" <caitlinb@broadcom.com>
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-----Original Message-----
From: dwmalone@cnri.dit.ie on behalf of David Malone
Sent: Fri 1/12/2007 4:01 AM
To: Gavin McCullagh
Cc: Rob Sherwood; Caitlin Bestler; tcpm@ietf.org
Subject: Re: [tcpm] DoS attack from misbehaving receivers
=20
> We implemented a basic attack of this recently before realising that =
others
> had been here first.  I've thrown up a few quick results here using a
> recent linux kernel on sender and receiver:

> 	http://www.hamilton.ie/gavinmc/drop_dupack_attack/

Note, while this was in the lab, we achieved similar results in the
wild. Using a clean Linux+Apache install over the Internet, we
convinced Apache to send a large file at 80Mbps to a host with a
1Mbps access link.  In light of some of the recent discussion on
Bugtraq, we probably don't really even need the large file.

We'd also arrived at basically the same fix as suggested by Rob.
Would it be interesting to see the fix written up as a draft, so
technical issues surrounding it can be identified? Certainly, some
care is required in the details of the fix to avoid introducing
other attacks on TCP. (It could fit naturally as an additional fix
section in draft-azcorra-tcpm-tcp-blind-ack-dos?)

	David.

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

Documenting the attack is definitely a good idea, and I agree
that it is essentially a blind ack attack (even though it is a=20
partial blindness, since *some* of the send segments reach
the attacker).

But I still don't see doing anything non-standard in the TCP
stack as a counter-measure.

Single flows of over 10 Mbits being sent to a stranger over
the wide internet can be easily flagged at many different=20
layers.

Keep in mind that *any* form of rate shaping will severely
impact this attack because the attacker will no longer be able
to predict the timing of tcp sequences.

Slightly random rate shaping has no impact on TCP, and
could easily be just as effective.





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From tcpm-bounces@ietf.org Sat Jan 13 05:59:56 2007
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> Documenting the attack is definitely a good idea, and I agree
> that it is essentially a blind ack attack (even though it is a 
> partial blindness, since *some* of the send segments reach
> the attacker).

Yes - indeed. This is kind of important, because it means that the
defences currently described in draft-azcorra-tcpm-tcp-blind-ack-dos
will not be effective.

> Single flows of over 10 Mbits being sent to a stranger over
> the wide internet can be easily flagged at many different 
> layers.

Sometimes this is not so easy - some people want to ship 10Mbps
streams to those that can receive them but do not want send them
to people for whom it will cause congestion (for example, people
providing mirror services). I guess it could be flagged by the
network, but few people want routers to send source quench messages
any more ;-)

To me it seems natural to address this at the TCP layer because it
is essentially an attack on TCP's congestion control mechanism.

> Keep in mind that *any* form of rate shaping will severely
> impact this attack because the attacker will no longer be able
> to predict the timing of tcp sequences.

There's no prediction involved in the attack that we were demoing
- it is entirely deterministic. We just send an ACK for every packet
we get, providing the sequence number is higher than the previous ACK
that we sent.

	David.

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From: Gavin McCullagh <gavin.mccullagh@nuim.ie>
Subject: Re: [tcpm] DoS attack from misbehaving receivers
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Hi,

On Fri, 12 Jan 2007, Caitlin Bestler wrote:

> But I still don't see doing anything non-standard in the TCP
> stack as a counter-measure.

Of course, if it were well thought-out enough to be made part of the
standard then it ceases to be non-standard :-)

> Single flows of over 10 Mbits being sent to a stranger over the wide
> internet can be easily flagged at many different layers.

I think expecting every potentially large-data TCP application to implement
rate-limiting is odd when the danger can only be understood at the
transport layer.  

We'd be expecting every sysadmin to make some guess what their cap should
be.  Should it be half of their uplink, a tenth, a hundredth?  Is it a
function also of the number of connections they expect today or even of the
number of attackers?  What if the attack is not on the sender but on some
narrower bottleneck between the sender and the attacker?

It's a workaround in some instances but I don't see how it could be a
solution to this problem.

> Keep in mind that *any* form of rate shaping will severely
> impact this attack because the attacker will no longer be able
> to predict the timing of tcp sequences.

Perhaps I've misunderstood you here, but I don't think this would prevent
the attack we have implemented.  What the attack does is simply to
acknowledge those packets which are received regardless of whether they are
in sequence.  There's no prediction at all.  As the attack progresses,
a constant number of acks is sent -- one for each packet received, but the
amount of data being acknowledged grows linearly (though some ack division
could probably improve on that).

	http://www.hamilton.ie/gavinmc/drop_dupack_attack/

In a sense "optimistic acking" is not really the best name for this
incarnation of the attack.  Perhaps better names might be "out of sequence
acking", "hole acking" or just plain "dishonest acking" to cover the
multitude.

> Slightly random rate shaping has no impact on TCP, and could easily be
> just as effective.

I'm not sure if you mean variation of MSS or of the congestion avoidance
increase?  Again, I'm often wrong but I don't think that's effective
against the above attack.

Gavin


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From tcpm-bounces@ietf.org Sat Jan 13 11:19:56 2007
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From: Rob Sherwood <capveg@cs.umd.edu>
To: Caitlin Bestler <caitlinb@broadcom.com>
Subject: Re: [tcpm] DoS attack from misbehaving receivers
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In addition to David and Gavin's comments:

On Fri, Jan 12, 2007 at 05:05:20PM -0800, Caitlin Bestler wrote:
> Single flows of over 10 Mbits being sent to a stranger over
> the wide internet can be easily flagged at many different 
> layers.

There is no requirement in the attack that any one flow would be large
(10MB or what have you) or in any way anomalous.  As discussed in the
paper, the attacker can simply create many small streams instead of few
large streams such that no intrusion detection system or rate limiting
queue is affected.  Because the ACK stream is used to determine when to
timeout (ACK clocking, forged TCP timestamps), when to send (incoming ACK
frees new segments), and how much to send (the number of new segments
freed), the attacker is in complete control of the connection, and can
shape traffic nearly arbitrarily.  So an intelligent attacker can shape
each connection to appear to be a perfectly "normal" connection to any
other layer then layer4, for any definition of normal.  That is why the
fix to this problem must be at the transmission layer.

> Keep in mind that *any* form of rate shaping will severely
> impact this attack because the attacker will no longer be able
> to predict the timing of tcp sequences.

I do not believe this to be true.  The attack described by David and Gavin
("lazy optack" in my paper) defeats this.  Additionally, the standard
optack attack defeats this when the attack simply ACKs less often (e.g.,
1 ack/500ms) but to more streams for the same total aggregate traffic.
This behavior, as described in the paper, is actually required in practice
to prevent ACKing ahead of sent data, because non-randomized rate TCP
streams naturally have sufficient entropy in the segment generation rate
that this is already a concern.

- Rob
.

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From tcpm-bounces@ietf.org Wed Jan 17 20:25:29 2007
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Hi.

As we mentioned in San Diego, the anti-spoof document is ready for a
second last call.  We're primarily checking to see that issues brought
up in the first WGLC have been addressed.  You can find much of that
discussion starting here:

http://www1.ietf.org/mail-archive/web/tcpm/current/msg01890.html

The draft is available from=20

ftp://ftp.ietf.org/internet-drafts/draft-ietf-tcpm-tcp-antispoof-05.txt

Comments (including positive ones, please) to the list.

This WGLC will end 2 Feb 2007 - a little more than 2 weeks from now.

--=20
Ted Faber
http://www.isi.edu/~faber           PGP: http://www.isi.edu/~faber/pubkeys.=
asc
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From tcpm-bounces@ietf.org Wed Jan 17 22:11:42 2007
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Date: Wed, 17 Jan 2007 22:08:08 -0500
From: Wesley Eddy <weddy@grc.nasa.gov>
To: Ted Faber <faber@ISI.EDU>
Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-tcp-antispoof-05.txt (Ends 2
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On Wed, Jan 17, 2007 at 05:24:41PM -0800, Ted Faber wrote:
> Hi.
>=20
> As we mentioned in San Diego, the anti-spoof document is ready for a
> second last call.  We're primarily checking to see that issues brought
> up in the first WGLC have been addressed.  You can find much of that
> discussion starting here:
>=20
> http://www1.ietf.org/mail-archive/web/tcpm/current/msg01890.html
>=20
> The draft is available from=20
>=20
> ftp://ftp.ietf.org/internet-drafts/draft-ietf-tcpm-tcp-antispoof-05.txt
>=20
> Comments (including positive ones, please) to the list.
>=20

Consider this a positive comment.  I've read both the current document
and some previous versions.  I think all the issues I noticed in
previous versions have been solved, and I think this is ready to publish
as an RFC.

I found the draft very helpful in understanding the aspects of the
problem and the gamut of solution approaches that TCPM has discussed.  I
hope other WGs that maintain well-established protocols follow this
example in documenting problems that are found over time and solution
approaches that are proposed.  Too often we only see a gamut of
solutions published, and no good description of the actual problem or
the engineering tradeoffs between the solutions.

--=20
Wesley M. Eddy
Verizon Federal Network Systems

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Subject: [tcpm] Alfred Hoenes: draft-ietf-tcpm-icmp-attacks-01 (etc.)
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Folks,

I received this review of the ICMP attacks draft.


>From: Alfred =3D?hp-roman8?B?SM5uZXM=3D?=3D <ah@tr-sys.de>
>Subject: draft-ietf-tcpm-icmp-attacks-01 (etc.)
>To: fernando@gont.com.ar
>Date: Wed, 10 Jan 2007 01:10:02 +0100 (MEZ)
>X-Mailer: ELM [$Revision: 1.17.214.3 $]
>
>Hello,
>after studying your I-D, draft-ietf-tcpm-icmp-attacks-01,
>I would like to send you e few comments pointing out (mostly)
>textual issues I found in that draft (and in a related note
>you have posted).
>
>After reading your exposition, I really wonder why these issues,
>and the countermeasures against, have not been discovered already
>many years ago -- such clear is the threat, und such logical seem
>the remedies!
>
>Here are the textual issues I found in the draft, documented as:
>
>    <original text>
>---
>    <proposed/changed text>
>
>I use change bars ('|' in column 1) and occasionally
>up/down pointing marker lines ('^^^'/'vvv') to emphasize
>the location of textual issues and/or proposed corrections.
>All snippits are taken preserving the original identation
>and formatting; modified text has been re-adjusted to match
>I-D / RFC formatting rules, where appropriate.
>
>
>(1)  Section 1:
>
>In the first paragraph:
>
>    ICMP [RFC0792] is a fundamental part of the TCP/IP protocol suite,
>    and is used mainly for reporting network error conditions.  However,
>    the current specifications do not recommend any kind of validation
>|  checks on the received ICMP error messages, thus allowing variety of
>    attacks against TCP [RFC0793] by means of ICMP, which include blind
>    connection-reset, blind throughput-reduction, and blind performance-
>    degrading attacks.  All of these attacks can be performed even being
>    off-path, without the need to sniff the packets that correspond to
>    the attacked TCP connection.
>---
>    ICMP [RFC0792] is a fundamental part of the TCP/IP protocol suite,
>    and is used mainly for reporting network error conditions.  However,
>    the current specifications do not recommend any kind of validation
>|  checks on the received ICMP error messages, thus allowing a variety
>    of attacks against TCP [RFC0793] by means of ICMP, which include
>    blind connection-reset, blind throughput-reduction, and blind
>    performance-degrading attacks.  All of these attacks can be performed
>    even being off-path, without the need to sniff the packets that
>    correspond to the attacked TCP connection.
>
>(2)  Section 2.2
>
>(2a)
>In the 4th paragraph, there's the first occurrence of an 'unsatisfied'
>reference tag (see marked line below):   [Touch-antispoof]
>This same tag recurs in the last sentence of Section 4.1, but it does
>*not* appear in the References sections.
>
>I guess (after browsing the I-D index) that it should be:
>             draft-ietf-tcpm-tcp-antispoof-05
>
>Please add the appropriate bibliographic entry to Section 10.2
>
>(2b)
>In the same paragraph, there's a typo (singular/plural mismatch):
>
>    Generally, the four-tuple required to perform these attacks is not
>|  known.  However, as discussed in [Watson] and [Touch-antispoof],
>    there are a number of scenarios (notably that of TCP connections
>    established between two BGP routers), in which an attacker may be
>    able to know or guess the four-tuple that identifies a TCP
>    connection.  In such a case, if we assume the attacker knows the two
>    systems involved in the TCP connection to be attacked, both the
>    client-side and the server-side IP addresses could be known or be
>    within a reasonable number of possibilities.  Furthermore, as most
>    Internet services use the so-called "well-known" ports, only the
>|  client port number might need to be guessed.  In such a scenario, an
>    attacker would need to send, in principle, at most 65536 packets to
>    perform any of the attacks described in this document.  However, as
>    most systems choose the port numbers they use for outgoing
>    connections from a subset of the whole port number space, the amount
>    of packets needed to successfully perform any of the attacks
>    discussed in this document could be further reduced.
>---
>    Generally, the four-tuple required to perform these attacks is not
>    known.  However, as discussed in [Watson] and [Touch-antispoof],
>    there are a number of scenarios (notably that of TCP connections
>    established between two BGP routers), in which an attacker may be
>    able to know or guess the four-tuple that identifies a TCP
>    connection.  In such a case, if we assume the attacker knows the two
>    systems involved in the TCP connection to be attacked, both the
>    client-side and the server-side IP addresses could be known or be
>    within a reasonable number of possibilities.  Furthermore, as most
>    Internet services use the so-called "well-known" ports, only the
>|  client port number might needs to be guessed.  In such a scenario, an
>    attacker would need to send, in principle, at most 65536 packets to
>    perform any of the attacks described in this document.  However, as
>    most systems choose the port numbers they use for outgoing
>    connections from a subset of the whole port number space, the amount
>    of packets needed to successfully perform any of the attacks
>    discussed in this document could be further reduced.
>
>(3)  Section 2.3
>
>In the first paragraph, there's a typo (missing "of"), and a sentence
>with two verbs.  I guess the "receives" should be deleted in favour of
>the "is":
>
>|  Section 5.2 of [RFC4301] describes the processing inbound IP Traffic
>    in the case of "unprotected-to-protected".  In the case of ICMP, when
>    an unprotected ICMP error message is received, it is matched to the
>    corresponding security association by means of the SPI (Security
>    Parameters Index) included in the payload of the ICMP error message.
>    Then, local policy is applied to determine whether to accept or
>    reject the message and, if accepted, what action to take as a result.
>    For example, if an ICMP unreachable message is received, the
>    implementation must decide whether to act on it, reject it, or act on
>    it with constraints.  Section 8 ("Path MTU/DF processing") discusses
>    the processing of unauthenticated ICMP "fragmentation needed and DF
>    bit set" (type 3, code 3) and ICMPv6 "Packet Too Big" (type 2, code
>|  0) messages when an IPsec implementation receives is configured to
>    process (vs. ignore) such messages.
>---
>|  Section 5.2 of [RFC4301] describes the processing of inbound IP
>    Traffic in the case of "unprotected-to-protected".  In the case of
>    ICMP, when an unprotected ICMP error message is received, it is
>    matched to the corresponding security association by means of the SPI
>    (Security Parameters Index) included in the payload of the ICMP error
>    message.  Then, local policy is applied to determine whether to
>    accept or reject the message and, if accepted, what action to take as
>    a result.  For example, if an ICMP unreachable message is received,
>    the implementation must decide whether to act on it, reject it, or
>    act on it with constraints.  Section 8 ("Path MTU/DF processing")
>    discusses the processing of unauthenticated ICMP "fragmentation
>    needed and DF bit set" (type 3, code 3) and ICMPv6 "Packet Too Big"
>|  (type 2, code 0) messages when an IPsec implementation is configured
>    to process (vs. ignore) such messages.
>
>(4)  Section 5.2
>
>I did not understand the last sentence in that (single) paragraph
>there, until I finally arrived at guesssing that the word "advice"
>perhaps should be deleted.
>
>                                                        [...].  Given the
>    hostile environments in which TCP currently operates in, and that
>|  advice ICMP provides an attack vector that is easier to exploit than
>    others (such as those discussed in [I-D.ietf-tcpm-tcpsecure]), we
>    believe that the improvements in TCP's resistance to these attacks
>    justify not taking the advice provided by the "SHOULDs" in [RFC1122].
>---
>                                                        [...].  Given the
>    hostile environments in which TCP currently operates in, and that
>|  ICMP provides an attack vector that is easier to exploit than others
>    (such as those discussed in [I-D.ietf-tcpm-tcpsecure]), we believe
>    that the improvements in TCP's resistance to these attacks justify
>    not taking the advice provided by the "SHOULDs" in [RFC1122].
>
>(5)  Section 5.2.1
>
>(5a)
>In the first paragraph, the wording should be improved:
>
>    An analysis of the circumstances in which ICMP messages that indicate
>|  hard errors may be received can shed some light to eliminate the
>    impact of ICMP-based blind connection-reset attacks.
>---
>    An analysis of the circumstances in which ICMP messages that indicate
>|  hard errors may be received can shed some light on how to eliminate
>    the impact of ICMP-based blind connection-reset attacks.
>
>(5b)
>In the discussion of the
>
>    ICMPv6 type 1 (Destination Unreachable), code 1 (communication with
>    destination administratively prohibited)
>
>there is a sentence that did not make sense to me; maybe it does not
>even say what you wanted to say :
>
>                    [...]  However, there is no reason to think that in
>       the same way this error condition appeared, it will not get solved
>       in the near term.  [...]
>
>I guess it was intended to state something like:
>
>                           In real life, this error condition will appear
>       very rarely; if it persists, it will be handled correctly after the
>       ensuing ACK timeout.
>
>(5b)
>In the paragraph after the five ICMP message type explanations (this is
>the 5th-to-last paragraph of the section), there are two sentences with
>two verbs each: "is" + "is" , and "are" + "indicate" , respectively:
>
>    Therefore, when following the reasoning explained above, TCPs in
>    synchronized states would treat all of the above messages as
>    indicating "soft errors", rather than "hard errors", and thus would
>|  not abort the corresponding connection upon receipt of them.  This is
>    policy is based on the premise that TCP should be as robust as
>    possible.  Reaction to these messages when they are meant for
>    connections in non-synchronized states could still remain as adviced
>    by [RFC1122], as we consider the attack window for connections in the
>    non-synchronized states is very small, and error messages received in
>|  these states are more likely indicate that the connection was opened
>    improperly [RFC0816].  Additionally, for the sake of robustness and
>    security, those implementations following the reasoning explained in
>    this section would not extrapolate ICMP errors across TCP
>    connections.
>---
>    Therefore, when following the reasoning explained above, TCPs in
>    synchronized states would treat all of the above messages as
>    indicating "soft errors", rather than "hard errors", and thus would
>|  not abort the corresponding connection upon receipt of them.  This
>    policy is based on the premise that TCP should be as robust as
>    possible.  Reaction to these messages when they are meant for
>    connections in non-synchronized states could still remain as adviced
>    by [RFC1122], as we consider the attack window for connections in the
>    non-synchronized states is very small, and error messages received in
>|  these states more likely indicate that the connection was opened
>    improperly [RFC0816].  Additionally, for the sake of robustness and
>    security, those implementations following the reasoning explained in
>    this section would not extrapolate ICMP errors across TCP
>    connections.
>
>(5c)
>In the subsequent paragraph, the above 'magic' sentence recurs in
>slightly modified form:
>
>    In case the received message were legitimate, it would mean that the
>    error condition appeared during the life of the corresponding
>|  connection.  However, in many scenarios there is no reason to think
>|  that in the same way this error condition appeared, it will not get
>|  solved in the near term.  [...]
>
>"there is no reason to think that ... it will not get solved ..."
>just contains too many negations.  This sentence might be replaced
>in a similar way as proposed above.
>
>(5d)
>Stepping ahead, in the next paragraph (3rd-to-last in the Section),
>the wording should preferrably be improved:
>
>|  One scenario in which a host would benefit from treating the so-
>    called ICMP "hard errors" as "soft errors" would be that in which the
>    packets that correspond to a given TCP connection are routed along
>    multiple different paths.  [...]
>---
>|  Another scenario in which a host would benefit from treating the so-
>    called ICMP "hard errors" as "soft errors" would be that in which the
>    packets that correspond to a given TCP connection are routed along
>    multiple different paths.  [...]
>
>(5e)
>The next paragraph deals with ignored error messages and out-of-band
>signalling of error conditions.  Therefore, "received error messages"
>should better be replaced by just "error conditions" :
>
>    It is interesting to note that, as ICMP error messages are
>    unreliable, transport protocols should not depend on them for correct
>    functioning.  In the event one of these messages were legitimate, the
>    corresponding connection would eventually time out.  Also,
>|  applications may still be notified asynchronously about the received
>|  error messages, and thus may still abort their connections on their
>    own if they consider it appropriate.
>---
>    It is interesting to note that, as ICMP error messages are
>    unreliable, transport protocols should not depend on them for correct
>    functioning.  In the event one of these messages were legitimate, the
>    corresponding connection would eventually time out.  Also,
>|  applications may still be notified asynchronously about the error
>|  conditions, and thus may still abort their connections on their own
>    if they consider it appropriate.
>
>[ In place of "error conditions", you might prefer "error condition",
>   "errors", or "error" . ]
>
>(6)  Section 5.2.3
>
>(6a)
>As in (5e) above (including the note), in the first paragraph:
>
>                 [...].  It should also be noted that applications may
>|  still be notified asynchronously about the received error messages,
>    and thus may still abort their connections on their own if they
>    consider it appropriate.
>---
>                 [...].  It should also be noted that applications may
>|  still be notified asynchronously about the error conditions, and thus
>    may still abort their connections on their own if they consider it
>    appropriate.
>
>(6b)
>And again, in the second paragraph:
>
>        [...].  Additionally, applications may still be notified
>|  asynchronously about the received error messages, and thus may still
>    abort their connections on their own if they consider it appropriate.
>---
>        [...].  Additionally, applications may still be notified
>|  asynchronously about the error conditions, and thus may still
>    abort their connections on their own if they consider it appropriate.
>
>(7)  Section 6.1
>
>At the end of the (single) paragraph, add the missing article, "an":
>
>|              [...].  The throughput achieved during attack might be a
>    little higher if a larger initial congestion window is in use
>    [RFC3390].
>---
>|              [...].  The throughput achieved during an attack might be
>    a little higher if a larger initial congestion window is in use
>    [RFC3390].
>
>(8)  Section 7.1
>
>I propose a textual enhancement in the 5th paragraph.
>IMHO, the referred to "IRQ ... rate" problem is only one face of the
>whole story.  Below, I substitute "protocol processing time" for that
>(you might also add it, as "IRQ ... rate and protocol processing time").
>
>                                                        [...].  On
>|  virtually all systems this will lead to an increase in the IRQ
>|  (Interrrupt ReQuest) rate, thus increasing processor utilization, and
>    degrading the overall system performance.
>---
>                                                        [...].  On
>|  virtually all systems this will lead to an increase in the protocol
>|  processing time, thus increasing processor utilization, and degrading
>    the overall system performance.
>
>(9)  Section 7.2
>
>(9a)
>In the classification performed:
>
>    To achieve both goals, we can divide the traditional PMTUD mechanism
>    into two stages: Initial Path-MTU Discovery, and Path-MTU Update.
>
>the latter apparently is intended to deal with PMTU reduction only,
>as can be seen from the subsequent discussion.
>
>IMHO, that has one serious drawback: it does not cover the "probing
>for PMTU increase" described in the third paragraph of Section 3 of
>RFC 1191, intended to discover effective PMTU introduced by routing
>changes.  I'm not sure whether this procedure is handled gracefully
>by your algorithm -- but I might err here; I just do not have enough
>spare time to delve into performing 'mental simulation' for that
>scenario now.  Please check.
>
>In any case, I would appreciate an extended discussion clearly
>covering this third case, throughout Section 7.2, and perhaps also
>an addition to Appendix A showing the behaviour during PMTU probing.
>
>(9b)
>In the 17th paragraph of Section 7.2, perhaps "everytime" should be
>replaced by "every time", according to the context.
>
>(10)  References
>
>(10a)  missing entry -- see (2a) above
>
>(10b)
>There's a typo in your own URI  [Ouch!] :
>
>    [ICMP-Filtering]
>               Gont, F., "Filtering of ICMP error messages",  http://
>               www.gont.com.ar/papers/filtering-icmp-error-messages.pdf.
>---
>    [ICMP-Filtering]
>               Gont, F., "Filtering of ICMP error messages",  http://
>               www.gont.com.ar/papers/filtering-of-icmp-error-messages.pdf.
>                                               ^^^
>BTW:
>... And in that memo, I found two small typos as well:
>
>(F1)  First bullet on page 1:
>       "on of"  -->  "one of"
>
>(F2)  First subordinate bullet of first bullet on page 2:
>       "intermmediate"  -->  "intermediate"
>
>(11)  Appendix B
>
>(11a)
>Repeated "far" <--> "for" ; missing punctuation
>
>    maxsizeacked
>       Variable holding the largest packet size (data, plus headers) that
>|     has so for been acked far this connection, as explained in
>|     Section 7.2
>---
>    maxsizeacked
>       Variable holding the largest packet size (data, plus headers) that
>|     has so far been acked for this connection, as explained in
>|     Section 7.2.
>
>    maxsizesent
>       Variable holding the largest packet size (data, plus headers) that
>|     has so for been sent far this connection, as explained in
>|     Section 7.2
>---
>    maxsizesent
>       Variable holding the largest packet size (data, plus headers) that
>|     has so far been sent for this connection, as explained in
>|     Section 7.2.
>
>    nsegrto
>       Variable holding the number of times this segment has timed out,
>|     as explained in Section 7.2
>---
>    nsegrto
>       Variable holding the number of times this segment has timed out,
>|     as explained in Section 7.2.
>
>    packet_size
>|     Variable holding the size of the IP datagram being sent
>---
>    packet_size
>|     Variable holding the size of the IP datagram being sent.
>
>(11b)
>In the pseudocode for
>
>    EVENT: Segment is received
>
>correct
>         if (pending_mesage)
>---
>         if (pending_message)
>
>(11c)
>The presentation of the pseudocode for
>
>    EVENT: ICMP "Packet Too Big" message is received
>
>suffers from the deep nesting and varying use of braces.
>[Note: That's perfectly o.k. from a syntax point of view (in 'C').]
>I propose to streamline the presentation to enhance the readability,
>by making it more compact and better show the structure of a single
>case switch (exactly one of the code blocks is to be executed in
>any case):
>
>         if (claimedtcpseq < SND.UNA || claimed_TCP_SEQ >=3D SND.NXT){
>              drop_message();
>|       }
>|
>|       else {
>|            if (claimedmtu >=3D maxsizesent || claimedmtu >=3D=
 current_mtu)
>                   drop_message();
>|
>|            else {
>|                 if (claimedmtu > maxsizeacked){
>                        adjust_mtu();
>                        current_mtu =3D claimedmtu;
>                        maxsizesent =3D MINIMUM_MTU;
>|                 }
>|
>|                 else {
>                        pending_message =3D 1;
>                        save_message();
>|                 }
>|            }
>         }
>---
>|       if (claimedtcpseq < SND.UNA || claimed_TCP_SEQ >=3D SND.NXT) {
>|           drop_message();
>|       } else if (claimedmtu >=3D maxsizesent ||
>|                  claimedmtu >=3D current_mtu) {
>|           drop_message();
>|       } else if (claimedmtu > maxsizeacked) {
>|           adjust_mtu();
>|           current_mtu =3D claimedmtu;
>|           maxsizesent =3D MINIMUM_MTU;
>|       } else {
>|           pending_message =3D 1;
>|           save_message();
>|       }
>
>[ This time, I also have tagged the lines that only had their
>   indentation changed, because that's the visual clue intended; I have
>   removed the redundant braces of type   else { if <statement> } ,
>   but added (arguably redundant) braces around the second instance of
>   drop_message();  just for symmetry;  the line break after '||' is
>   introduced only to accomodate for the line length restriction. ]
>
>(12)  Appendix E
>
>(12a)
>Apparently there is something wrong with E.1 and E.2 :
>
>E.1.  Changes from draft-gont-tcpm-icmp-attacks-05       <---+
>E.2.  Changes from draft-ietf-tcpm-icmp-attacks-00       <---+
>E.3.  Changes from draft-gont-tcpm-icmp-attacks-04
>E.4.  Changes from draft-gont-tcpm-icmp-attacks-03
>E.5.  Changes from draft-gont-tcpm-icmp-attacks-02
>E.6.  Changes from draft-gont-tcpm-icmp-attacks-01
>E.7.  Changes from draft-gont-tcpm-icmp-attacks-00
>
>Shouldn't the sequence of the subsections correspond to the history,
>with the  draft-ietf-*-00  being the latest, listed first?
>
>I do not know whether just the headlines are exchanged, or the full
>sub-sections are misoredered.  Please check.
>
>(12b)  typo in current E.2 :
>
>|  o  Added an specific section on IPsec (Section 2.3)
>---
>|  o  Added a specific section on IPsec (Section 2.3)
>
>(12b)  garbage in E.6 :
>
>|  o  Added Section on Acknowledgement number checking"/>
>---
>|  o  Added Section on Acknowledgement number checking
>
>
>Best regards,
>   Alfred H=CEnes.
>
>--
>
>+------------------------+--------------------------------------------+
>| TR-Sys Alfred Hoenes   |  Alfred Hoenes   Dipl.-Math., Dipl.-Phys.  |
>| Gerlinger Strasse 12   |  Phone: (+49)7156/9635-0, Fax: -18         |
>| D-71254  Ditzingen     |  E-Mail:  ah@TR-Sys.de                     |
>+------------------------+--------------------------------------------+

--
Fernando Gont
e-mail: fernando@gont.com.ar || fgont@acm.org
PGP Fingerprint: 7809 84F5 322E 45C7 F1C9 3945 96EE A9EF D076 FFF1






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From tcpm-bounces@ietf.org Wed Jan 24 14:49:15 2007
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Date: Wed, 24 Jan 2007 11:43:26 -0800
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Subject: [tcpm] WGLC draft-ietf-behave-tcp-04 starts now
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BEHAVE, TCPM:

We are starting a two-week working group last call in the BEHAVE working
group for a TCP-related document, draft-ietf-behave-tcp-04, which describes
how to NAT TCP traffic.  This document is expected to become a BCP.

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

     Abstract:
     This document defines a set of requirements for NATs that
     handle TCP that would allow many applications, such as
     peer-to-peer applications and on-line games, to work
     consistently.  Developing NATs that meet this set of
     requirements will greatly increase the likelihood that
     these applications will function properly.
    
Please send comments to the authors (CC'd) or behave@ietf.org.

WGLC on this document ends on Wednesday, 7-Feb-2007.

-Dan Wing
 chair, BEHAVE working group

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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-03.txt
	Pages		: 13
	Date		: 2007-1-25
	
This document discusses the problem of long delays between connection
   establishment attempts that may arise in a number of scenarios,
   including one in which dual stack nodes that have IPv6 enabled by
   default are deployed in IPv4 or mixed IPv4 and IPv6 environments.
   Additionally, this document describes a non-standard, but widely
   implemented modification to TCP's reaction to ICMP "soft errors" that
   can help overcome this problem.

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On Wed, Jan 17, 2007 at 05:24:41PM -0800, Ted Faber wrote:
> Hi.
>=20
> As we mentioned in San Diego, the anti-spoof document is ready for a
> second last call.  We're primarily checking to see that issues brought
> up in the first WGLC have been addressed.  You can find much of that
> discussion starting here:
>=20
> http://www1.ietf.org/mail-archive/web/tcpm/current/msg01890.html
>=20
> The draft is available from=20
>=20
> ftp://ftp.ietf.org/internet-drafts/draft-ietf-tcpm-tcp-antispoof-05.txt
>=20
> Comments (including positive ones, please) to the list.
>=20
> This WGLC will end 2 Feb 2007 - a little more than 2 weeks from now.

Just a reminder: this WGLC has one week left.  I've only seen one
message about the draft.  Passing a WGLC requires some reaction from the
group.  If you're interested in seeing this work done and believe that
this document represents a reasonable consensus, please speak up.  And,
of course, if you see problems with the draft, also speak up.

--=20
Ted Faber
http://www.isi.edu/~faber           PGP: http://www.isi.edu/~faber/pubkeys.=
asc
Unexpected attachment on this mail? See http://www.isi.edu/~faber/FAQ.html#=
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From tcpm-bounces@ietf.org Fri Jan 26 13:33:24 2007
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To: TCP Maintenance and Minor Extensions WG <tcpm@ietf.org>
From: David Borman <david.borman@windriver.com>
Date: Fri, 26 Jan 2007 12:33:29 -0600
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Subject: [tcpm] RFC 1323: Timestamps option
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One of the topics that has been discussed in the past for the  
revision of RFC 1323 is to relax the requirement on when to send the  
Timestamps option.  I'd like to come to some resolution on this issue.

Current:
The Timestamps option is negotiated during the initial SYN exchange.   
If both sides support it, then every packet in the connection has to  
have the Timestamps option.

Option #1
---------
The proposed change is to relax the requirement that Timestamps have  
to be negotiated in the initial SYN, and instead allow them to be  
enabled later on in the connection.  At the time 1323 was originally  
written, there was a big concern for legacy TCP stacks that didn't  
handle unknown options in the middle of the connection, but were able  
to properly ignore unknown options during the SYN exchange.  But that  
was a long time ago, and unknown options in the middle of a TCP  
connection *shouldn't* be as big of an issue anymore.

The rules for enabling Timestamps mid-stream would be:
1) If you receive a Timestamps option mid-stream, you can enable  
Timestamps.  From then on, every packet sent will need to contain a  
Timestamps option.
2) If you want to enable Timestamps mid-stream, you can only do so on  
a data packet.  You record the sequence number of that data packet,  
and include the Timestamps option on every packet from then on, until  
you receive an ACK for that sequence number.  If at any time you  
receive a packet with the Timestamps option, then they are enabled  
for the rest of the connection.  If you do not receive a packet with  
a Timestamps option by the time you get the ACK for that initial  
sequence number, then the other side has not enabled Timestamps and  
you must stop sending Timestamps for the duration of the connection.

The restriction on only being able to initiate Timestamps on a data  
packet is that they are the only packets that are delivered reliably,  
so determining whether or not the other side supports them is  
deterministic by getting the ACK for that data.


Option #2
---------
This is similar to Option #1, but not quite the same.  In this case,  
the Timestamps would be negotiated in the initial SYN as before, but  
then no additional packets would have Timestamps options.  At any  
point during the connection, either side can enable the use of  
Timestamps just by starting to send them.  Since we already know that  
both sides support them, once you start sending them you never stop.   
And if you every receive one, you are then required to also send them  
for the duration of the connection.

The question then is, how do you negotiate Timestamps without turning  
them on?  My suggestion would be to send a Timestamps option with the  
special format TSval=0,TSecr=0.  If you receive this in a SYN, then  
the request is to negotiate knowledge of the Timestamps option.  The  
returning SYN,ACK would also need to contain TSval=0,TSecr=0 to  
complete the negotiation.  If you are connecting to a site that  
doesn't support this feature but does support Timestamps, it would  
respond with TSval=xxx,TSecr=0.  At that point, Timestamps are now  
enabled.  The downside of this is that now the originator has started  
their Timestamps values at 0, and must continue from there with their  
timestamps; they can't just suddenly jump to some arbitrary value  
because that could break PAWs.


Option #3
---------
Do nothing, and leave Timestamps alone as they are currently  
defined.  You negotiate them in the SYN, and then include them in  
every packet from then on.

One of the issues with Timestamps is that the TSecr value is defined  
to be valid anytime the ACK bit is set.  This works well with the  
initial SYN negotiation, but when you enable Timestamps midstream,  
the initiator has to send a Timestamps option with an invalid TSecr  
value, but the ACK bit will be set.  That's the case for both of the  
previous options.

So why should we change things?  My recollection was that the primary  
motivation was to conserve option space, to avoid having to put in  
the extra 12 bytes of TCP options on every packet until we get to the  
point where the sequence space might wrap, and then turn them on so  
that we can get PAWS to protect the rest of the connection.  But I'm  
not sure I buy that argument.  One of the things that you lose by  
deferring enabling Timestamps is that you don't get the RTT  
measurements that Timestamps provide.  Without Timestamps, using the  
typical BSD algorithm of timing one segment and waiting for an ACK,  
and adding in the Karn algorithm that you have to invalidate any  
measurements if you had to retransmit in that window, means that at  
best, you'll get one measurement per RTT.  At worst, you can get in a  
state where you never get a valid RTT measurement.  When you use the  
Timestamps option, the worst case is that you'll only get one  
measurement per RTT, and the best case is that you'll get multiple  
valid samples,

My viewpoint
------------
Right now, I'm inclined to leave the Timestamps option alone; you  
negotiate it during the SYN exchange, and then include it on every  
packet.  I'm willing to be convinced otherwise, but now I've placed a  
stake in the ground, and the mailing list will have to decide if  
there is sufficient reason to change the behavior.

			-David Borman


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From tcpm-bounces@ietf.org Fri Jan 26 14:10:55 2007
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From: Sally Floyd <sallyfloyd@mac.com>
Subject: Re: [tcpm] RFC 1323: Timestamps option
Date: Fri, 26 Jan 2007 11:09:59 -0800
To: David Borman <david.borman@windriver.com>
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David -

I don't have any feedback about *when* to negotiate timestamps, but I 
do have some feedback
about how the RTO should be estimated when timestamps are used.  Just 
as something
to include in a revision of RFC 1323...

Regards,
- Sally
http://www.icir.org/floyd/

Here is some old email of mine on this:

> To: mallman@icir.org
> cc: "Ethan Blanton" <eblanton@cs.purdue.edu>,
>     "Aaron Falk" <falk@isi.edu>, "Ted Faber" <faber@isi.edu>,
>     touch@isi.edu, "Scott Bradner" <sob@harvard.edu>
> Bcc:
> From: Sally Floyd <floyd@icir.org>
> Subject: Re: Mark Allman: would this fly?
> Date: Tue, 09 Aug 2005 13:45:18 -0700
> Sender: floyd@cougar.icir.org
>
> Mark -
>
> >> It is worth noting that if timestamps are being used, with the
> >> currently-specified time constant for estimating the average RTT and
> >> variance, then the first thing that needs to be done is to change 
> the
> >> time constant so that the average RTT and the variance aren't being
> >> estimated over a small fraction of a window of data (as is possible
> >> when timestamps are used with current parameters).  (Unless this has
> >> already been fixed in recent standards, and I just missed it...)
> >
> >I don't think this has been fixed.  And, I don't see this potential 
> i-d
> >as the place to fix it.  I wanted this to be an alternative response 
> to
> >spurious timeouts, not some general RTO change.  Does that make sense?
>
> It makes sense.  Except that if in fact the average and variance are
> being estimated over a small fraction of a window size, because
> the TCP connection is using timestamps and has a moderate-to-large
> congestion window, then the estimated variance might be very small
> (because it is measured over RTTs calculated essentially from the
> most recent N packets, for N < cwnd, and the burstiness for rtts for
> the most recent fraction of a cwnd might be quite small).
>
> So in that situation, increasing the multiple of the variance might
> have almost no effect on the RTO, because the variance might be
> unrealistically small when calculated only over this small time
> scale.  And changing the weight for estimating the rtt average and
> variance to make sure that the average RTT and variance are estimated
> from RTTs calculated over a period of several rtts, might make your
> technique much more powerful, by making the calculated variance
> much more interesting and useful.  Worth noting, at any rate.  IMHO.
>
> - Sally


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This note on the TCP SYN flooding draft is forwarded with permission
from its author.  All of the suggested changes look right to me, and
will show up in the next update of the draft.


----- Forwarded message from Alfred H=F6nes <ah@tr-sys.de> -----

From: Alfred H=F6nes <ah@tr-sys.de>
Subject: draft-ietf-tcpm-syn-flood-01
To: weddy@grc.nasa.gov
Date: Thu, 25 Jan 2007 10:15:56 +0100 (MEZ)

Hello,
after studying the Internet-Draft authored by you,
   draft-ietf-tcpm-syn-flood-01 ,
I'd like to submit a few comments, mostly addressing textual
flaws I found in that memo.

Generally, the presentation of the topic is very welcomed,
it's over-due!
Also, technical coverage seems to be complete and reasonable.
Therefore, no specific objections!

The items below address the remaining nits I found in the draft;
they are presented in textual order, with one exception, at the end.
To give more context, sometimes I quote larger blocks of text
literally and show the replacement proposed using the shorthand
notation:

   <original draft text>
---
   <modified text>

I use change bars ('|' in column 1) and occasionally
up/down pointing marker lines ('^^^'/'vvv') to emphasize
the location of textual issues and/or proposed corrections.
Modified text has been re-adjusted to match RFC formatting
rules, where appropriate.  I also try to accomodate published
RFC-Ed policy on style and punctuation etc. in my proposals.


(1)  Section 1, last paragraph: improper wording

   This majority of this document consists of three sections.  [...]
---
   The majority of this document consists of three sections.  [...]


(2)  Section 2.2, first paragraph: punctuation

   As described in RFC 793, a TCP implementation may allow the LISTEN
|  state to be entered with either all, some. or none of the pair of IP
   addresses and port numbers specified by the application.  [...]
---                                         ^
   As described in RFC 793, a TCP implementation may allow the LISTEN
|  state to be entered with either all, some, or none of the pair of IP
   addresses and port numbers specified by the application.  [...]


(3)  Section 2.2, 4th paragraph: word omission (?)

                                                [...].  In other systems
   that implement less complex TCP algorithms and options, the overhead
   may be less, although typically exceeds 280 bytes [SKK+97].
---
                                                [...].  In other systems
   that implement less complex TCP algorithms and options, the overhead
|  may be less, although it typically exceeds 280 bytes [SKK+97].


(4)  Section 3.7: two typos

|  Firewall-baed tactics may also be used to defend end-hosts from SYN
   flooding attacks.  The basic concept is to offload the connection
|  establishment proceedures onto a firewall that screens connection
   attempts until they are completed and then proxies them back to
   protected end hosts.  [...]
---
|  Firewall-based tactics may also be used to defend end-hosts from SYN
   flooding attacks.  The basic concept is to offload the connection
|  establishment procedures onto a firewall that screens connection
   attempts until they are completed and then proxies them back to
   protected end hosts.  [...]


(5)  Section 4, 2nd paragraph: typo

   Among end host modifications, the SYN cache and SYN cookie approaches
|  seem to be the only viable techniques discoverd.  Increasing the
   backlog and reducing the SYN-RECEIVED timer are measurably
   problematic.  [...]
---
   Among end host modifications, the SYN cache and SYN cookie approaches
|  seem to be the only viable techniques discovered.  Increasing the
   backlog and reducing the SYN-RECEIVED timer are measurably
   problematic.  [...]


(6)  Section 4, 4th paragraph: grammar

   In 1997, NetBSD incorporated a modified version of Borman's code.
   Two notable differences from the original code stem from the decision
|  to use of the cache by default (for all connections).  This implied
   the need to perform retransmissions for SYN-ACKs, and to use larger
   structures to keep more complete data.  [...]
---
   In 1997, NetBSD incorporated a modified version of Borman's code.
   Two notable differences from the original code stem from the decision
|  to use the cache by default (for all connections).  This implied the
   need to perform retransmissions for SYN-ACKs, and to use larger
   structures to keep more complete data.  [...]


(7)  Appendix A, 4th paragraph: grammar/typo
                                                    vvvvvvvvvvvvvvvvvvv
|  With SYN cookies enabled, a host will be able to maintain responsive
   even when under a SYN flooding attack.  [...]
---
|  With SYN cookies enabled, a host will be able to remain responsive
   even when under a SYN flooding attack.  [...]

(Or use: "maintain responsiveness" -- I prefer the shorter form above.)


(8)  Appendix A, 6th paragraph: word omission
                                                                v
|  The best description of the exact SYN cookie algorithms is in part of
   an email from Bernstein, that is archived on the web site (notice it
   does not set the top five bits from the counter modulo 32, as the
   previous description did, but instead uses 29 bits from the second
   MD5 operation and 3 bits for the index into the MSS table;
   establishing the secret values is also not discussed):
---                                                             vvv
|  The best description of the exact SYN cookie algorithms is in a part
   of an email from Bernstein, that is archived on the web site (notice
   it does not set the top five bits from the counter modulo 32, as the
   previous description did, but instead uses 29 bits from the second
   MD5 operation and 3 bits for the index into the MSS table;
   establishing the secret values is also not discussed):


And last, but not least:

(9)  Section 6 - Acknowledgements

IMHO, the seminal work of D.J. Bernstein deserves a note in Section 6,
in particular for its 'contribution' in Appendix A.


Best regards,
  Alfred H=F6nes.

--=20

+------------------------+--------------------------------------------+
| TR-Sys Alfred Hoenes   |  Alfred Hoenes   Dipl.-Math., Dipl.-Phys.  |
| Gerlinger Strasse 12   |  Phone: (+49)7156/9635-0, Fax: -18         |
| D-71254  Ditzingen     |  E-Mail:  ah@TR-Sys.de                     |
+------------------------+--------------------------------------------+

----- End forwarded message -----

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To: David Borman <david.borman@windriver.com>
From: Mark Allman <mallman@icir.org>
Subject: Re: [tcpm] RFC 1323: Timestamps option 
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[obviously, without co-chair hat]

Hi Dave-

I am not going to vote yet.  I want to try to understand what you write
a little more ...

> One of the things that you lose by deferring enabling Timestamps is
> that you don't get the RTT measurements that Timestamps provide.
> Without Timestamps, using the typical BSD algorithm of timing one
> segment and waiting for an ACK, and adding in the Karn algorithm that
> you have to invalidate any measurements if you had to retransmit in
> that window, means that at best, you'll get one measurement per RTT.
> At worst, you can get in a state where you never get a valid RTT
> measurement.  When you use the Timestamps option, the worst case is
> that you'll only get one measurement per RTT, and the best case is
> that you'll get multiple valid samples,

So, 

  + I agree that getting into a state whereby you never get an RTT
    measurement is Not Good.

    (Although, one could think of ways to avoid this and we might also
    decide this is pathological.)

  + Why is the "best case" that you'll get multiple samples per RTT?  It
    seems to me that the underlying assumption is that "more is better",
    but I am not sure it is.  We have done some analysis that suggests
    more samples are not better.  I don't want to say our analysis is
    the last word here, but I'd be interested in hearing arguments for
    taking more samples [using the current RTO estimator] per RTT being
    a clear win.

    I think this is an important point in choosing a path.  I.e., if
    multiple RTT measurements matter then that might suggest a different
    approach than if multiple RTTs buy you little/nothing.

    Ref:

        Mark Allman, Vern Paxson.  On Estimating End-to-End Network Path
        Properties.  Proceedings of the ACM SIGCOMM Technical Symposium,
        Cambridge, MA, September 1999.
        http://www.icir.org/mallman/papers/estimation.ps

allman



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




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From tcpm-bounces@ietf.org Fri Jan 26 14:43:19 2007
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From: David Borman <david.borman@windriver.com>
Subject: Re: [tcpm] RFC 1323: Timestamps option
Date: Fri, 26 Jan 2007 13:43:47 -0600
To: Sally Floyd <sallyfloyd@mac.com>
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Hi Sally,

Yes, thanks.  That is one of the other open issue with 1323.  Over  
the next week or so I'm planning on sending out individual emails for  
each of the other issues, so that they can be discussed individually,  
rather than trying to discuss them all in one big email thread.  But  
since you brought it up, let's begin this thread now. :-)

For the RFC, at a minimum we need to add a description of the  
problem, and say that implementors should take into account the more  
frequent timestamp values.  I don't think we should trying to tell  
them exactly how to address the issue, but it'd be nice to have a  
reasonably example that could be easily implemented and achieve the  
desired effect.

So, my idea is to use a two-tier approach.  As without Timestamps,  
you record the sequence number to begin measuring the RTT.  Until you  
get the ACK for that sequence number, you take all the measurements  
from the Timestamps options and average them together.  Once the ACK  
for the sequence number returns, you take the averaged value and use  
that in the traditional RTO calculations, and reset the sequence  
number and averaged value.  There are probably better algorithms, but  
this one is simple and should be fairly easy to implement, and would  
produce better results than the situation today where all the  
Timestamps are fed into the RTO calculation without adjusting for the  
increased frequency of the measurements.

		-David Borman

On Jan 26, 2007, at 1:09 PM, Sally Floyd wrote:

> David -
>
> I don't have any feedback about *when* to negotiate timestamps, but  
> I do have some feedback
> about how the RTO should be estimated when timestamps are used.   
> Just as something
> to include in a revision of RFC 1323...
>
> Regards,
> - Sally
> http://www.icir.org/floyd/
>
> Here is some old email of mine on this:
>
>> To: mallman@icir.org
>> cc: "Ethan Blanton" <eblanton@cs.purdue.edu>,
>>     "Aaron Falk" <falk@isi.edu>, "Ted Faber" <faber@isi.edu>,
>>     touch@isi.edu, "Scott Bradner" <sob@harvard.edu>
>> Bcc:
>> From: Sally Floyd <floyd@icir.org>
>> Subject: Re: Mark Allman: would this fly?
>> Date: Tue, 09 Aug 2005 13:45:18 -0700
>> Sender: floyd@cougar.icir.org
>>
>> Mark -
>>
>> >> It is worth noting that if timestamps are being used, with the
>> >> currently-specified time constant for estimating the average  
>> RTT and
>> >> variance, then the first thing that needs to be done is to  
>> change the
>> >> time constant so that the average RTT and the variance aren't  
>> being
>> >> estimated over a small fraction of a window of data (as is  
>> possible
>> >> when timestamps are used with current parameters).  (Unless  
>> this has
>> >> already been fixed in recent standards, and I just missed it...)
>> >
>> >I don't think this has been fixed.  And, I don't see this  
>> potential i-d
>> >as the place to fix it.  I wanted this to be an alternative  
>> response to
>> >spurious timeouts, not some general RTO change.  Does that make  
>> sense?
>>
>> It makes sense.  Except that if in fact the average and variance are
>> being estimated over a small fraction of a window size, because
>> the TCP connection is using timestamps and has a moderate-to-large
>> congestion window, then the estimated variance might be very small
>> (because it is measured over RTTs calculated essentially from the
>> most recent N packets, for N < cwnd, and the burstiness for rtts for
>> the most recent fraction of a cwnd might be quite small).
>>
>> So in that situation, increasing the multiple of the variance might
>> have almost no effect on the RTO, because the variance might be
>> unrealistically small when calculated only over this small time
>> scale.  And changing the weight for estimating the rtt average and
>> variance to make sure that the average RTT and variance are estimated
>> from RTTs calculated over a period of several rtts, might make your
>> technique much more powerful, by making the calculated variance
>> much more interesting and useful.  Worth noting, at any rate.  IMHO.
>>
>> - Sally
>


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Subject: Re: [tcpm] RFC 1323: Timestamps option
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On Fri, 26 Jan 2007, David Borman wrote:

> One of the topics that has been discussed in the past for the
> revision of RFC 1323 is to relax the requirement on when to send the
> Timestamps option.  I'd like to come to some resolution on this issue.
>
> Current:
> The Timestamps option is negotiated during the initial SYN exchange.
> If both sides support it, then every packet in the connection has to
> have the Timestamps option.

... snip ...

>
> My viewpoint
> ------------
> Right now, I'm inclined to leave the Timestamps option alone; you
> negotiate it during the SYN exchange, and then include it on every
> packet.  I'm willing to be convinced otherwise, but now I've placed a
> stake in the ground, and the mailing list will have to decide if
> there is sufficient reason to change the behavior.

The real problem is a zero configuration issue:  for the 100 million or so US
home users on DSL (or worse, compressing modems), timestamps offer little
gain and often substantial penalty.  For the 10 Million(?) US university
users, timestamps are necessary for good performance.  Who do you think the
manufacturers cater to? The simple answer is the home user, but many current
systems use some sort of heuristic (e.g. is the interface faster than 100
Mb/s?).

The current situation forces us to try to educate 9.99 million non-network
savvy users how to inspect and tweak their kernels.

I want ZERO config from at least 9.6 kbs to 9 Gbps.  Some form of late
timestamp negotiation is required before this can become the default.

(BTW Japan and Europe, with true high speed to the home might change the
equation quite a bit.  This path leads to some ugly market positions.)

Thanks,
--MM--

P.S.  The default WSCALE for MS Vista and mainline Linux is now 8 and 7
respectively for large memory systems, so the harder part of the zero config
problem has already been solved.  These correspond to 16 and 8 MByte TCP
buffers. Roll out is going to take a while, but that will give people a chance
to notice how the traffic might affect the network.  QoS anyone?

Thanks,
--MM--
-------------------------------------------
Matt Mathis      http://www.psc.edu/~mathis
Work:412.268.3319    Home/Cell:412.654.7529
-------------------------------------------
Evil is defined by mortals who think they know
"The Truth" and use force to apply it to others.


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Subject: Re: [tcpm] RFC 1323: Timestamps option
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Hi David,

I'm inclined for option #3, too.

Some comments inline....


>Option #2
>---------
[....]
>The question then is, how do you negotiate Timestamps without turning
>them on?  My suggestion would be to send a Timestamps option with the
>special format TSval=0,TSecr=0.  If you receive this in a SYN, then
>the request is to negotiate knowledge of the Timestamps option.  The
>returning SYN,ACK would also need to contain TSval=0,TSecr=0 to
>complete the negotiation.  If you are connecting to a site that
>doesn't support this feature but does support Timestamps, it would
>respond with TSval=xxx,TSecr=0.  At that point, Timestamps are now
>enabled.  The downside of this is that now the originator has started
>their Timestamps values at 0, and must continue from there with their
>timestamps; they can't just suddenly jump to some arbitrary value
>because that could break PAWs.

IIRC, there are some implementations that randomize the TCP sequence 
number, but use monotonically increasing timestamp generator.
Those implementations would cause, the BSD hack of allowing the 
establishment of a new connection if the ISN of the new incarnation 
of the conenction is larger than the last seq number seen on the old 
connection to fail fail.
IIRC, Linux also takes into account the Timestamp. Thus, if you 
randomize the ISN, but use a monotonically increasing timestamp 
generator, the hack would still succeed.

However, option #2 above would break this.


[....]
>So why should we change things?  My recollection was that the primary
>motivation was to conserve option space, to avoid having to put in
>the extra 12 bytes of TCP options on every packet until we get to the
>point where the sequence space might wrap, and then turn them on so
>that we can get PAWS to protect the rest of the connection.

I assume this means that one would turn timestamps on when starting a 
bulk transfer on the connection (?).

If so I'm not sure the space we'd be saving would be worth the 
additional complexity: For most application protocols usually 
employed for bulk transfers, the bulk transfer is preceded by only a 
few packets (to establish the TCP connection and send the application 
request). So option space would be saved in only a few segments.

Am I missing something?

Also, in order to enable timestamps midstream, I guess applications 
should perform a call to setsockopt() or the like? If so, this 
require specification of the corresponding socket option, would 
require applications to be modified, and we would also requier us to 
agree on "what the option should default to", etc.

So, unless some strong reason is raised for going with option #1 or 
option #2, I would stick with option #3.

Thanks,

-- 
Fernando Gont
e-mail: fernando@gont.com.ar || fgont@acm.org
PGP Fingerprint: 7809 84F5 322E 45C7 F1C9 3945 96EE A9EF D076 FFF1





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From tcpm-bounces@ietf.org Fri Jan 26 15:05:11 2007
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From: David Borman <david.borman@windriver.com>
Subject: Re: [tcpm] RFC 1323: Timestamps option 
Date: Fri, 26 Jan 2007 14:05:34 -0600
To: mallman@icir.org
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You've probably already seen my reply to Sally, but I'll clarify a  
few things:

On Jan 26, 2007, at 1:42 PM, Mark Allman wrote:

>   + Why is the "best case" that you'll get multiple samples per  
> RTT?  It
>     seems to me that the underlying assumption is that "more is  
> better",
>     but I am not sure it is.  We have done some analysis that suggests
>     more samples are not better.  I don't want to say our analysis is
>     the last word here, but I'd be interested in hearing arguments for
>     taking more samples [using the current RTO estimator] per RTT  
> being
>     a clear win.
>
>     I think this is an important point in choosing a path.  I.e., if
>     multiple RTT measurements matter then that might suggest a  
> different
>     approach than if multiple RTTs buy you little/nothing.

Perhaps "best case" was the wrong term.  Being guaranteed at least  
one good RTT measurement per RTT is a good thing.  The current RTO  
estimator is based on getting one sample per RTT, no matter how much  
data flows during that time.  As you know, when you start using all  
the samples from multiple Timestamps options, if you just feed them  
into the current RTO estimator it can seriously skew the results,  
especially if you are getting lots of samples per RTT.  In my reply  
to Sally, I suggested taking the average of all the Timestamps  
received during one RTT, and feeding that into the RTO estimator.  An  
alternative would be to take the largest sample received during the  
RTT interval, and feed that into the estimator.  That'd be even  
easier to implement.  Do you have any thoughts as to which would be  
better?

Another alternative would be to only use the sample from the  
Timestamps option in the ACK for the sequence number that is being  
timed, i.e. back to one sample per RTT.  But that is still a win over  
the old method, because with Timestamps, we know that the measurement  
will always be accurate, even if there was a retransmission during  
the RTT.

But I totally agree, just feeding in the more frequent samples from  
the Timestamps into the RTO estimator is not a good thing.

		-David Borman



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From tcpm-bounces@ietf.org Fri Jan 26 16:09:46 2007
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Subject: Re: [tcpm] RFC 1323: Timestamps option 
In-Reply-To: Your message of "Fri, 26 Jan 2007 12:33:29 CST."
	<018977E8-C9F4-42E2-A877-95330F65E7D3@windriver.com> 
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> The rules for enabling Timestamps mid-stream would be:
> 1) If you receive a Timestamps option mid-stream, you can enable  
> Timestamps.  From then on, every packet sent will need to contain a  
> Timestamps option.

Based on some of the behaviour noted in:

	http://www.caida.org/publications/papers/2005/fingerprinting/

It seems some implementations have already effectively chosen Option #1.  

	David.

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From tcpm-bounces@ietf.org Fri Jan 26 20:42:22 2007
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Subject: Re: [tcpm] RFC 1323: Timestamps option 
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> You've probably already seen my reply to Sally, but I'll clarify a
> few things:

Yes - I just read it (although, I am a bit behind on the thread, so
apologies if some of the stuff below has been said).  But, I think it
is orthogonal to my concern.  Likely, I am not stating things well.

Sally's observation (and, she is not the only one to make it over the
years) is that the current TCP RTO algorithm keeps history in terms of
the *number* of samples taken.  So, if you take two RTT samples per RTT
as opposed to one then you are keeping half as much history in *time
units*.  I think we all agree with this and that it should be addressed
**when a TCP connection is taking more than one RTT sample per RTT**.
Exactly how we address it is a discussion we can have (averaging samples
into one per RTT that is fed to the estimator, taking the max sample per
RTT, adjusting the EWMA gains based on the cwnd, etc.).  This problem
manifests itself *when* we take multiple samples per RTT.  What I am
asking about is *whether* we need multiple samples per RTT.

It seems to me that your notes assume somehow that more than one sample
per RTT is useful.  Why?  Or, 

> Being guaranteed at least one good RTT measurement per RTT is a good
> thing.

Why?

(I don't mean to be flip.  Surely one good RTT sample per RTT is a fine
goal.  The "why?" is in response to the "guaranteed" part of your
statement.) 

Clearly we want an evolution of the RTO to take into account changing
RTTs and the variance in the RTT.  So, we don't want to use a static
number and we do not want to take a single sample and then stop
(although, actually, this latter does not perform too bad).

Let me try to put this a different way... what is the problem with the
non-TS one sample / RTT sampling technique?  You have noted a couple of
things:

  + That we might not be able to get *any* RTT sample.

  + That Karn's algorithm means that during some RTTs we will not get a
    sample.

  + (Are there others?)

The first of these is a clear problem that using the TS option from the
beginning can solve (although, a small tweak to retransmit SYNs with
timestamps even if the original did not have a TS may well fix this).
I do wonder how much I should care about this case.  It seems like it
will be exceedingly rare to get into this situation and further even if
we do I am not sure I care about performance (if that means including a
12-byte blob on every packet sent for this express reason).

The second case above is for sure true.  But, the question is does it
matter at all?  In other words, what is the impact on the RTO of getting
a sample during these RTTs as opposed to not?  

I ask these questions to test the assumptions.  My own thinking goes
like this .... Timestamps are essentially a general disambiguation
mechanism.  There is a clear need for some form of disambiguation when
the sequence space wraps too quickly.  That mechanism does not have to
be timestamps, but they work fine.  However, such a disambiguation
scheme is not needed when the sequence space cannot wrap too quickly.

RFC 1323 also laid out a couple reasons for using a disambiguation
mechanism for RTT timing.  One of those is because the ambiguity leads
to no RTT samples in some situations (i.e., when we retransmit).  The
other item is the [what I would consider] conjecture that more RTT
samples help the RTO.  In the time since RFC 1323 was published I think
there is some evidence that says that the need for disambiguation for
RTT sampling is thin.  Maybe I am biased by my own work.  Our work is
one data point.  What I am really looking for is other work / evidence /
experience that makes the case that we need disambiguation for RTT
sampling in anything but corner cases and then what do we gain from this
disambiguation?

allman




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From tcpm-bounces@ietf.org Fri Jan 26 20:48:01 2007
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To: Fernando Gont <fernando@gont.com.ar>
From: Mark Allman <mallman@icir.org>
Subject: Re: [tcpm] RFC 1323: Timestamps option 
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> I assume this means that one would turn timestamps on when starting a
> bulk transfer on the connection (?).
> 
> If so I'm not sure the space we'd be saving would be worth the
> additional complexity: For most application protocols usually employed
> for bulk transfers, the bulk transfer is preceded by only a few
> packets (to establish the TCP connection and send the application
> request). So option space would be saved in only a few segments.
> 
> Am I missing something?
> 
> Also, in order to enable timestamps midstream, I guess applications
> should perform a call to setsockopt() or the like? If so, this require
> specification of the corresponding socket option, would require
> applications to be modified, and we would also requier us to agree on
> "what the option should default to", etc.

My view is that the application would not have to be involved.  The app
is not involved in whether to use timestamps or not right now.  (Setting
the socket buffer quite large could be seen as a proxy of this, I
suppose.)  In any case, I don't think you need to worry about knowing a
file is big or not.  In order to need TS for PAWS you have to send 4GB
of data (i.e., wrap).  It should be easy enough to turn TS on after 3GB
or something.  (It really isn't the amount of data being sent, it is the
data rate that PAWS is protecting against and so you could come up with
a better test than simply testing the number of bytes sent---my point is
that without actually wrapping we don't have to worry about TS for
PAWS.)

allman




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From tcpm-bounces@ietf.org Sat Jan 27 07:23:02 2007
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Subject: Re: [tcpm] RFC 1323: Timestamps option
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A difficulty with the suggested enabling mechanism for Option #2 is
that a major TCP stack already (mis)behaves in a similar way for current
TCP connections.

> Option #2
> ---------
...
> The question then is, how do you negotiate Timestamps without turning  
> them on?  My suggestion would be to send a Timestamps option with the  
> special format TSval=0,TSecr=0.  If you receive this in a SYN, then  
> the request is to negotiate knowledge of the Timestamps option.  The  
> returning SYN,ACK would also need to contain TSval=0,TSecr=0 to  
> complete the negotiation.

The Windows TCP stack currently replies SYN,ACK TSval=0,TSecr=0 to a
regular SYN TSval=xxx,TSecr=0.

So if you send SYN TSval=0,TSecr=0, and get back SYN,ACK TSval=0,TSecr=0
you may be talking to a TCP stack that understands the new enabling
semantics, or a Windows TCP stack that does not.

You could argue this does not really matter, as the Windows TCP stack
will go on using Timestamps (I presume), immediately enabling it on the
fly under the new semantics, so in the end all is fairly well (except
Windows TCP may be sent a few segments with no Timestamps that it does
not expect, until on the fly enabling is completed).  However I don't
think this confusing situation is desirable.

I think Windows TCP behaves in this way because it does not create a
full TCP state block until after SYN,ACK processing, so has nowhere
to record the initial TSvals.  If we are updating RFC1323, I wonder if
this Windows behaviour should be explicitly allowed, since in practice
we have to allow for it now, and it has some merit.

Trying to find legitimacy for the Windows TCP behaviour, it could
be said that 3.2 of RFC1323 wasn't entirely clear in this regard.
It says "When TSecr is not valid, its value must be zero." - I can
conceive of this being understood (out of context) as meaning: if you
do not have a valid TSecr available for some reason, then send a zero;
an interpretation that would permit the Windows TCP behaviour, and make
the suggested enabling mechanism ambiguous.

	Richard
-- 
Richard Wendland				richard@codeburst.co.uk

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Just to echo some of the comments from Matt and Mark, I've always felt =
that the overhead of 12 bytes per packet (and ACK) was pretty unbearable =
and usually recommend for turning off timestamps if that is possible in =
an application.

Clearly, we need some solution for PAWS, and I think the comments about =
turning on timestamps based on rate or volume of data sent in those =
cases make good sense, so I'd be in favor of that.

For the cases where you don't need timestamps for PAWS, why not simply =
allow a host to transmit a timestamp option whenever it wants to measure =
the RTT?  Then hosts would have the freedom to use timestamps at =
whatever rate they feel is best.  Having read some of the comments about =
frequency of measurements, it occurs to me that it might actually be =
better to do one measurement per fixed unit of time (say, once per =
second) rather than once per packet or once per RTT.  I don't think =
there would be any reason to think that variability of network delay =
would be correlated to RTT; it seems much more likely to be correlated =
to fixed constants in the network (like routing protocol update =
intervals) or fixed constants in the real world (like business hours).

Of course, from my point of view the advantage to this is that you'd be =
adding 24 bytes per second (or whatever the chosen interval is) to the =
data transmitted rather than 18 bytes per packet (taking into account =
delayed ACK) which is more than 1% additional overhead for fairly =
nebulous benefit.

I realize this suggestion would probably require a more significant =
change to the operation of the timestamp option, but I think it would be =
worthwhile.

--J


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From tcpm-bounces@ietf.org Mon Jan 29 00:09:41 2007
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Date: Mon, 29 Jan 2007 13:08:24 +0800
From: Kacheong Poon <kacheong.poon@sun.com>
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To: David Borman <david.borman@windriver.com>
Subject: Re: [tcpm] RFC 1323: Timestamps option
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David Borman wrote:

> Option #3
> ---------
> Do nothing, and leave Timestamps alone as they are currently defined. 
> You negotiate them in the SYN, and then include them in every packet
> from then on.


As an implementor, I'd vote for Option #3.  If the current behavior
is not broken, don't fix it.  Modifying it after so many years of
deployment may cause many troubles.  IMHO, the benefit of modifying
the handling, which is saving 12 bytes, is minor when compared to
the effort of diagnosing why a connection is not working.  If we
want to fix the overhead and other issues, such as auto-tuning, I'd
rather to have a new option or other methods than modifying the
current handling.

I'd like to have several clarifications on timestamps handling.

1. I think it is not stated explicitly that once timestamps option
   is enabled, it should always be used for a connection.  I have
   encountered TCP stack which stops sending timestamps option in
   the middle of a connection.  Could this be made explicitly in
   the update?  I also suggest putting a note on how to handle such
   behavior.  For example, once one side stops sending the option,
   the peer will treat it as if the option is never negotiated and
   also stops using it.

2. As described in CERT VU#637934, there are some inconsistencies
   in RFC 1323 and the not "published" RFC 1323bis.  This creates
   some "variants" of algorithms.  Since this update will obsolete
   both documents, please check the CERT report on the issues and
   clarify it on the update.

3. It is stated that RST should not have a timestamps option.  I
   think this is really not needed.  And it only causes extra check
   in the code not to do that.  For example, an app does an abortive
   close, the code needs to remove the timetstamps option from the
   connection to send the RST.  I don't think it causes any harm if
   the RST has the option.




-- 

						K. Poon.
						kacheong.poon@sun.com


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On 2007-1-18, at 3:24, ext Ted Faber wrote:
> This WGLC will end 2 Feb 2007 - a little more than 2 weeks from now.

I did my AD review during WGLC to speed up things a bit.

Overall, this is a very solid document that is ready for publication.

There are some minor boilerplate and spelling/grammar issues. Since I  
know Joe uses Word, turn on spelling and grammar checks :-) If there  
are more substantial comments that would require a revision now, fix  
these nits, too, otherwise they can wait until AUTH48.

 >    At 1 Mbps, 57,000 (40
 >    byte) RSTs would take over 50 minutes to transmit, and, as noted
 >    earlier, most current connections are fairly brief by comparison.

   57000*8*40 = 18,240,000 bits need 18.24 seconds with 1Mbps

Lars



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From tcpm-bounces@ietf.org Mon Jan 29 10:56:21 2007
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From: Gavin McCullagh <gavin.mccullagh@nuim.ie>
Subject: Re: [tcpm] RFC 1323: Timestamps option
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Hi,

On Fri, 26 Jan 2007, David Borman wrote:

> But I totally agree, just feeding in the more frequent samples from                                                           
> the Timestamps into the RTO estimator is not a good thing.                                                                    

Indeed. Naturally, it is possible to adjust the RTO estimation to be able
to deal with receiving estimates more often, or even irregularly. Quite a
few OSes (eg Darwin, I think?) use a RTO timer (and time stamps) that ticks
every half a second, and so effectively rounds up to the nearest 500ms.  I
guess this shows that it doesn't matter too much what you do, as long as
you give a reasonable overestimate.

It is conceivable that people might want to use timestamps for things other
than RTO estimation. Some of the high-speed TCP variants want more RTT
samples (we're playing with one that expects an RTT sample per ACK at the
moment).  I don't think any of the suggested new behaviours would cause
problems with these uses, but it may be worth keeping in mind while
updating RFC 1323.

As an example of a more exotic use of the time stamp option, the most
recent version of FreeBSD's syncookies stashes information in the
timestamp:

        http://lists.freebsd.org/pipermail/freebsd-net/2006-September/011651.html

Gavin


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From tcpm-bounces@ietf.org Mon Jan 29 11:16:49 2007
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Hi, all,

I received some detailed corrections from Alfred Hoenes, noting the
issues below, as well as the suggestion to include a sentence on NATs.

Would it be more useful to make these corrections before or after the WG
sends the document on? (and can they be done without yet another WGLC?)

Joe

Lars Eggert wrote:
> On 2007-1-18, at 3:24, ext Ted Faber wrote:
>> This WGLC will end 2 Feb 2007 - a little more than 2 weeks from now.
>=20
> I did my AD review during WGLC to speed up things a bit.
>=20
> Overall, this is a very solid document that is ready for publication.
>=20
> There are some minor boilerplate and spelling/grammar issues. Since I
> know Joe uses Word, turn on spelling and grammar checks :-) If there ar=
e
> more substantial comments that would require a revision now, fix these
> nits, too, otherwise they can wait until AUTH48.
>=20
>>    At 1 Mbps, 57,000 (40
>>    byte) RSTs would take over 50 minutes to transmit, and, as noted
>>    earlier, most current connections are fairly brief by comparison.
>=20
>   57000*8*40 =3D 18,240,000 bits need 18.24 seconds with 1Mbps
>=20
> Lars
>=20
>=20
>=20
> -----------------------------------------------------------------------=
-
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
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--=20
----------------------------------------
Joe Touch
Sr. Network Engineer, USAF TSAT Space Segment


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Subject: Re: [tcpm] AD review of draft-ietf-tcpm-tcp-antispoof-05.txt
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On 2007-1-29, at 18:16, ext Joe Touch wrote:
> Would it be more useful to make these corrections before or after  
> the WG
> sends the document on? (and can they be done without yet another  
> WGLC?)

I'd be better to fix these before the document goes to the IESG.  
Editorial fixes don't require another WGLC.

Lars



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From tcpm-bounces@ietf.org Wed Jan 31 10:15:04 2007
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Dear all,

A long time ago, at the San Diego IETF meeting in August 2004,
I presented an idea called "Corruption Notification Options"
(a proposal which is some sort of a refined specification of
the TCP HACK idea) to this group.

The group's feedback was that this is useless, as errors
occur in such a clustered fashion that you wouldn't see
any packets with an intact header but erroneous payload
(which is the only situation where the scheme would
yield any benefit). I think that it was Gorry Fairhurst
who said this.

I figured that the only convincing way to prove him
wrong is to actually do a real-life test. I did, and
proved him right  :)  that is, disabling checksums for
parts of packets doesn't yield much of a benefit in
WiFi networks, where it seems that frames are delivered
in an all-or-nothing fashion.

Doing this test requires patching the device driver (to
disable the link layer CRC), and above all, finding a good
student - all of this took quite some time, but now it's
over, and the documentation can be found at:
http://www.welzl.at/research/projects/corruption/
(bottom of the page)

I learned my lesson, and thought I'd share it with you -
but this result only concerns WiFi. I'll keep this as a
low-priority "background" project and may take a closer
look at other link layers one day... in any case, that
topic still interests me, and I can't believe that it
wouldn't be possible to get a benefit out of such a scheme.


I'm sending this note to TCPM, TSVWG and DCCP because there
are similar mechanisms there (DCCP: the Data Checksum option; SCTP:
http://www.ietf.org/internet-drafts/draft-stewart-sctp-pktdrprep-06.txt
although I just noticed that this is apparently not a WG item...
still, it's SCTP related).

This note also goes to ICCRG, because I think that this should
be the place for related discussions (in fact Lachlan Andrew
will give us a related talk at our upcoming meeting:
http://www1.tools.ietf.org/group/irtf/trac/wiki/Agenda )


Cheers,
Michael


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Michael Welzl wrote:
> Dear all,
>=20
> A long time ago, at the San Diego IETF meeting in August 2004,
> I presented an idea called "Corruption Notification Options"
> (a proposal which is some sort of a refined specification of
> the TCP HACK idea) to this group.
>=20
> The group's feedback was that this is useless, as errors
> occur in such a clustered fashion that you wouldn't see
> any packets with an intact header but erroneous payload
> (which is the only situation where the scheme would
> yield any benefit). I think that it was Gorry Fairhurst
> who said this.

There was a separate issue - you need to *know* the header is intact.
For TCP, there's no separate header checksum to provide that
information. I.e., this might be possible with UDP-lite, but isn't with
UDP or TCP. This might not affect the SCTP or DCCP variants, depending
on what their checksums cover.

Joe

--=20
----------------------------------------
Joe Touch
Sr. Network Engineer, USAF TSAT Space Segment


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From tcpm-bounces@ietf.org Wed Jan 31 11:13:08 2007
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Subject: Re: [tcpm] Separate header checksums and WiFi
From: Michael Welzl <michael.welzl@uibk.ac.at>
To: Joe Touch <touch@ISI.EDU>
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On Wed, 2007-01-31 at 16:58, Joe Touch wrote:
> Michael Welzl wrote:
> > Dear all,
> > 
> > A long time ago, at the San Diego IETF meeting in August 2004,
> > I presented an idea called "Corruption Notification Options"
> > (a proposal which is some sort of a refined specification of
> > the TCP HACK idea) to this group.
> > 
> > The group's feedback was that this is useless, as errors
> > occur in such a clustered fashion that you wouldn't see
> > any packets with an intact header but erroneous payload
> > (which is the only situation where the scheme would
> > yield any benefit). I think that it was Gorry Fairhurst
> > who said this.
> 
> There was a separate issue - you need to *know* the header is intact.
> For TCP, there's no separate header checksum to provide that

Well, that's what my proposal added  :)


> information. I.e., this might be possible with UDP-lite, but isn't with
> UDP or TCP. This might not affect the SCTP or DCCP variants, depending
> on what their checksums cover.

It will affect the DCCP Data Checksum, which is just the
same as my TCP proposal. The SCTP proposal is a little
different.

Cheers,
Michael


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Michael Welzl spake unto us the following wisdom:
> I figured that the only convincing way to prove him
> wrong is to actually do a real-life test. I did, and
> proved him right  :)  that is, disabling checksums for
> parts of packets doesn't yield much of a benefit in
> WiFi networks, where it seems that frames are delivered
> in an all-or-nothing fashion.

I have to applaud this move.  All too often we only talk about the
things we did that *worked*, and the same questions about things that
any number of people would tell you doesn't work come up over and
over.  For this particular topic, there is now a citation, so the next
guy won't have to go try it himself.  Thank you.

Ethan

--=20
The laws that forbid the carrying of arms are laws [that have no remedy
for evils].  They disarm only those who are neither inclined nor
determined to commit crimes.
		-- Cesare Beccaria, "On Crimes and Punishments", 1764

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Michael Welzl wrote:
> On Wed, 2007-01-31 at 16:58, Joe Touch wrote:
=2E..
>> There was a separate issue - you need to *know* the header is intact.
>> For TCP, there's no separate header checksum to provide that
>=20
> Well, that's what my proposal added  :)

Did your experiment measure that aspect? That's the key question - how
many packets have recoverable headers...

Joe

--=20
----------------------------------------
Joe Touch
Sr. Network Engineer, USAF TSAT Space Segment


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From tcpm-bounces@ietf.org Wed Jan 31 12:14:03 2007
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Subject: Re: [tcpm] Separate header checksums and WiFi
From: Michael Welzl <michael.welzl@uibk.ac.at>
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On Wed, 2007-01-31 at 17:54, Joe Touch wrote:
> Michael Welzl wrote:
> > On Wed, 2007-01-31 at 16:58, Joe Touch wrote:
> ...
> >> There was a separate issue - you need to *know* the header is intact.
> >> For TCP, there's no separate header checksum to provide that
> > 
> > Well, that's what my proposal added  :)
> 
> Did your experiment measure that aspect? That's the key question - how
> many packets have recoverable headers...

Yes! It's a sad, sad story (well, for me it was  :)   ).
See the tables on pages 62ff of
http://www.welzl.at/research/projects/corruption/index.html

To give you an example, in the measurement on page 61,
we had approx. 479k received packets, of which 178 were
received with errors and 14 had an intact header checksum.

That's the approximate order of magnitude... tens or
hundreds, if not less, out of hundreds of thousands.

By the way, I'd like to point out that all the credit
for this measurement goes to the master student who
did it, Mattia Rossi - he really put a lot of effort
into this. BTW, he's now in Australia (Perth), taking
a break, and soon looking for a job there AFAIK;
his email address is (hopefully still working?)
mattia.rossi@uibk.ac.at and I'm cc'ing him.

Cheers,
Michael


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From tcpm-bounces@ietf.org Wed Jan 31 12:14:56 2007
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Beyond what Ethan is suggesting, this is useful to write up more fully
and publish, e.g., at Globecom or ICC. Negative results, as Ethan notes,
are critical to the community as a whole.

Joe

Ethan Blanton wrote:
> Michael Welzl spake unto us the following wisdom:
>> I figured that the only convincing way to prove him
>> wrong is to actually do a real-life test. I did, and
>> proved him right  :)  that is, disabling checksums for
>> parts of packets doesn't yield much of a benefit in
>> WiFi networks, where it seems that frames are delivered
>> in an all-or-nothing fashion.
>=20
> I have to applaud this move.  All too often we only talk about the
> things we did that *worked*, and the same questions about things that
> any number of people would tell you doesn't work come up over and
> over.  For this particular topic, there is now a citation, so the next
> guy won't have to go try it himself.  Thank you.
>=20
> Ethan
>=20
>=20
>=20
> -----------------------------------------------------------------------=
-
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www1.ietf.org/mailman/listinfo/tcpm

--=20
----------------------------------------
Joe Touch
Sr. Network Engineer, USAF TSAT Space Segment


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From tcpm-bounces@ietf.org Wed Jan 31 12:17:41 2007
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Subject: Re: [tcpm] Separate header checksums and WiFi
From: Michael Welzl <michael.welzl@uibk.ac.at>
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I agree, and am happy to contribute what I have!!!!!

(which is: the results, but not the time)


On Wed, 2007-01-31 at 18:14, Joe Touch wrote:
> Beyond what Ethan is suggesting, this is useful to write up more fully
> and publish, e.g., at Globecom or ICC. Negative results, as Ethan notes,
> are critical to the community as a whole.
> 
> Joe
> 
> Ethan Blanton wrote:
> > Michael Welzl spake unto us the following wisdom:
> >> I figured that the only convincing way to prove him
> >> wrong is to actually do a real-life test. I did, and
> >> proved him right  :)  that is, disabling checksums for
> >> parts of packets doesn't yield much of a benefit in
> >> WiFi networks, where it seems that frames are delivered
> >> in an all-or-nothing fashion.
> > 
> > I have to applaud this move.  All too often we only talk about the
> > things we did that *worked*, and the same questions about things that
> > any number of people would tell you doesn't work come up over and
> > over.  For this particular topic, there is now a citation, so the next
> > guy won't have to go try it himself.  Thank you.
> > 
> > Ethan
> > 
> > 
> > 
> > ------------------------------------------------------------------------
> > 
> > _______________________________________________
> > tcpm mailing list
> > tcpm@ietf.org
> > https://www1.ietf.org/mailman/listinfo/tcpm
-- 
Michael Welzl <michael.welzl@uibk.ac.at>
University of Innsbruck


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