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Date: Thu, 01 Feb 2007 11:44:04 +0100
From: Francesco Vacirca <francesco@net.infocom.uniroma1.it>
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To: Michael Welzl <michael.welzl@uibk.ac.at>
Subject: Re: [tcpm] Separate header checksums and WiFi
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Some link layers use a strongest FEC to protect header. E.g. in some 
UMTS coding scheme the link layer employs a 1/3 codification for RLC 
header, whereas the payload can use a different scheme (e.g. from 4/5 to 
1/3)... Maybe it could be applied also to TCP. Note that this can 
decrease the goodput in case of non lossy links... obviously it depends 
on the ratio between useful bits and transmitted bits.

In the 802.11 standard some part of the packet (MAC header) is sent with 
a different rate to be more protected against channel impairments and 
also for compatibility purposes. A cross layer approach could adopt low 
rate also for TCP header (also IP obviously)... but I do not think that 
the benefits are more than disadvantages.

One more thing... in case of Michael experiments, are the packet losses 
on the channel due to SNR fluctuations or due to MAC collisions?
In the second case, it is quite normal that the whole packet 
(header+payload) is corrupted.

Francesco

Michael Welzl wrote:
> 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

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From tcpm-bounces@ietf.org Thu Feb 01 05:45:03 2007
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below the answer from Michael:


On Thu, 2007-02-01 at 11:24, Francesco Vacirca wrote:
> Some link layers use a strongest FEC to protect header. E.g. in some 
> UMTS coding scheme the link layer employs a 1/3 codification for RLC 
> header, whereas the payload can use a different scheme (e.g. from 4/5 to 
> 1/3)... Maybe it could be applied also to TCP. Note that this can 
> decrease the goodput in case of non lossy links... obviously it depends 
> on the ratio between useful bits and transmitted bits.
> 
> In the 802.11 standard some part of the packet (MAC header) is sent with 
> a different rate to be more protected against channel impairments and 
> also for compatibility purposes. A cross layer approach could adopt low 
> rate also for TCP header (also IP obviously)... but I do not think that 
> the benefits are more than disadvantages.
> 
> One more thing... in case of Michael experiments, are the packet losses 
> on the channel due to SNR fluctuations or due to MAC collisions?
> In the second case, it is quite normal that the whole packet 
> (header+payload) is corrupted.

They are generally due to SNR fluctuations, by transmitting
between two notebooks in ad hoc mode (the only way that
disabling the CRC worked for us). I remember that Mattia
also did a quick check with one more notebook, to see if
MAC influences the result, but it didn't seem to play a role
(I don't remember if this test made it into the document in
the end... I think not).

It would be interesting to know why errors occur in the
fashion that we saw; we measured, but don't really have
an explanation. Perhaps it's the PHY coding.

Anyway, for those of you in who didn't see my previous
email and are confused, this link should explain it:
http://www.welzl.at/research/projects/corruption/index.html

Cheers,
Michael

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On Fri, Jan 26, 2007 at 09:47:42AM -0800, Ted Faber wrote:
> 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.
>=20
> 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.

One more heads up.  Today (2 Feb 2007) is the last day of this WGLC.
If you've been thinking hard or meaning to get to this, please send
something today.

Thanks.

--=20
Ted Faber
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From tcpm-bounces@ietf.org Fri Feb 02 14:50:10 2007
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From: Pekka Savola <pekkas@netcore.fi>
To: Ted Faber <faber@ISI.EDU>
Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-tcp-antispoof-05.txt (Ends 2
	Feb 2007)
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On Fri, 2 Feb 2007, Ted Faber wrote:
>>> 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.
>
> One more heads up.  Today (2 Feb 2007) is the last day of this WGLC.
> If you've been thinking hard or meaning to get to this, please send
> something today.

(As I already said earlier, I believe this is a document that should 
be IETF Last Called even if doing so is not required by RFC2026.)

I've had some comments on earlier versions of the draft.  I've now 
checked those.  The document isn't as explicit as I'd have liked 
particularly in section 3.2.1, but I find it mostly acceptable.  I 
object to the conclusion of section 3.2.1 which was not updated based 
on the latest discussion:

   As a result, address filtering is not a local solution that can be
    deployed to protect communicating pairs, but rather relies on a
    distributed infrastructure of trusted gateways filtering forged
    traffic where it enters the network.  It is not feasible for local,
    incremental deployment, and relies heavily on distributed
    cooperation.  Although useful to reduce the load of spoofed traffic,
    it is insufficient to protect particular connections from attack
    [29].

Specifically the second sentence seems somewhat assumptive of the 
communication patterns and usefulness thereof for the endpoints, and 
the resulting conclusion in the third sentence is also assumptive. 
Anyone will agree that address filtering is not a _complete_ solution 
for every communication a random host might want to do, but whether it 
protects some particular connections from attack depends on how those 
connections are situated related to the address filtering border(s) 
[and what filtering has been configured there] and where the attacker 
is (as is now slightly expanded in the 2nd paragraph).

Earlier, I suggested the following replacement text (for -02):

    As a result, address filtering can be deployed locally but it only
    protects communications where both parties are local.  Therefore, the
    larger the address filtering border is, the more extensive the protection
    is.  This is a particularly useful means of protection for specific kinds
    of sessions which are known to be local.  However, as address filtering
    is performed in the network by trusted gateways, the users not know
    whether the filtering exists or not.  Hence, the users cannot rely on it
    in general and other mechanisms for protection must be used as well.
    Still, address filtering is very useful to protect specific kinds of
    communications (e.g., network management, most router-to-router
    communication, etc.).

But a smaller delta to the current text might also be OK even if it is 
less explicit, for example:

    As a result, address filtering is not a local solution that can be
    deployed to protect communicating pairs, but rather relies on a
    distributed infrastructure of trusted gateways filtering forged
    traffic where it enters the network.  It is not feasible as a
    general solution due to the lack universal deployment, but
    may be applicable to connections among those inside the protected
    border in some scenarios.  Applying filtering also reduces
    the load of spoofed traffic [29].


As another comment, Section 3.2.1 also refers to RFC3682 section 5.1 
for TTL-security discussion, and also mentions BITW tunnels. 
draft-ietf-rtgwg-rfc3682bis would be a better reference here as the 
security properties of TTL-based security and particularly use with 
tunnels has been greatly expanded.

-- 
Pekka Savola                 "You each name yourselves king, yet the
Netcore Oy                    kingdom bleeds."
Systems. Networks. Security. -- George R.R. Martin: A Clash of Kings

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Hi, all,

Pekka Savola wrote:
> On Fri, 2 Feb 2007, Ted Faber wrote:
>>>> 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.
>>
>> One more heads up.  Today (2 Feb 2007) is the last day of this WGLC.
>> If you've been thinking hard or meaning to get to this, please send
>> something today.
> 
> (As I already said earlier, I believe this is a document that should be 
> IETF Last Called even if doing so is not required by RFC2026.)

I'm not sure every document wouldn't fall into that category; I disagree 
that this particular document should be treated uniquely.

> I've had some comments on earlier versions of the draft.  I've now 
> checked those.  The document isn't as explicit as I'd have liked 
> particularly in section 3.2.1, but I find it mostly acceptable.  I 
> object to the conclusion of section 3.2.1 which was not updated based on 
> the latest discussion:
> 
>   As a result, address filtering is not a local solution that can be
>    deployed to protect communicating pairs, but rather relies on a
>    distributed infrastructure of trusted gateways filtering forged
>    traffic where it enters the network.  It is not feasible for local,
>    incremental deployment, and relies heavily on distributed
>    cooperation.  Although useful to reduce the load of spoofed traffic,
>    it is insufficient to protect particular connections from attack
>    [29].
> 
> Specifically the second sentence seems somewhat assumptive of the 
> communication patterns and usefulness thereof for the endpoints, and the 
> resulting conclusion in the third sentence is also assumptive. Anyone 
> will agree that address filtering is not a _complete_ solution for every 
> communication a random host might want to do, but whether it protects 
> some particular connections from attack depends on how those connections 
> are situated related to the address filtering border(s) [and what 
> filtering has been configured there] and where the attacker is (as is 
> now slightly expanded in the 2nd paragraph).

I entirely agree that filtering works in some cases, depending on the 
specifics of where it is deployed and what the nature of the attack is. 
For example, if you're in control of all the distributed locations where 
you depend on filtering, or the topology is such that they're not very 
distributed, things get simpler. However, that doesn't seem a useful 
nuance to point out, IMO.

> Earlier, I suggested the following replacement text (for -02):
> 
>    As a result, address filtering can be deployed locally but it only
>    protects communications where both parties are local.  Therefore, the
>    larger the address filtering border is, the more extensive the 
> protection
>    is.  This is a particularly useful means of protection for specific 
> kinds
>    of sessions which are known to be local.  However, as address filtering
>    is performed in the network by trusted gateways, the users not know
>    whether the filtering exists or not.  Hence, the users cannot rely on it
>    in general and other mechanisms for protection must be used as well.
>    Still, address filtering is very useful to protect specific kinds of
>    communications (e.g., network management, most router-to-router
>    communication, etc.).
> 
> But a smaller delta to the current text might also be OK even if it is 
> less explicit, for example:
> 
>    As a result, address filtering is not a local solution that can be
>    deployed to protect communicating pairs, but rather relies on a
>    distributed infrastructure of trusted gateways filtering forged
>    traffic where it enters the network.  It is not feasible as a
>    general solution due to the lack universal deployment, but
>    may be applicable to connections among those inside the protected
>    border in some scenarios.  Applying filtering also reduces
>    the load of spoofed traffic [29].

I agree with the second sentence, which might be useful to include. As 
to the last sentence, this document focuses on endpoint mechanisms for 
defense, and doesn't address upstream load reduction.

> As another comment, Section 3.2.1 also refers to RFC3682 section 5.1 for 
> TTL-security discussion, and also mentions BITW tunnels. 
> draft-ietf-rtgwg-rfc3682bis would be a better reference here as the 
> security properties of TTL-based security and particularly use with 
> tunnels has been greatly expanded.

Thanks for pointing that out. Since the ID hasn't been published yet, 
it's probably useful to cite both documents on that point.

Joe

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


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From tcpm-bounces@ietf.org Sat Feb 03 01:50:26 2007
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From: Pekka Savola <pekkas@netcore.fi>
To: Joe Touch <touch@ISI.EDU>
Subject: Re: [tcpm] WGLC for draft-ietf-tcpm-tcp-antispoof-05.txt (Ends 2
	Feb 2007)
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On Fri, 2 Feb 2007, Joe Touch wrote:
>>    As a result, address filtering is not a local solution that can be
>>     deployed to protect communicating pairs, but rather relies on a
>>     distributed infrastructure of trusted gateways filtering forged
>>     traffic where it enters the network.  It is not feasible for local,
>>     incremental deployment, and relies heavily on distributed
>>     cooperation.  Although useful to reduce the load of spoofed traffic,
>>     it is insufficient to protect particular connections from attack
>>     [29].
...
>>  But a smaller delta to the current text might also be OK even if it is
>>  less explicit, for example:
>>
>>     As a result, address filtering is not a local solution that can be
>>     deployed to protect communicating pairs, but rather relies on a
>>     distributed infrastructure of trusted gateways filtering forged
>>     traffic where it enters the network.  It is not feasible as a
>>     general solution due to the lack universal deployment, but
>>     may be applicable to connections among those inside the protected
>>     border in some scenarios.  Applying filtering also reduces
>>     the load of spoofed traffic [29].
>
> I agree with the second sentence, which might be useful to include. As to the 
> last sentence, this document focuses on endpoint mechanisms for defense, and 
> doesn't address upstream load reduction.

I don't care about the last sentence, and it could be removed or 
reworded.  It was just an attempt to rephrase the last sentence in the 
original (above) so that reference [29] is preserved.

-- 
Pekka Savola                 "You each name yourselves king, yet the
Netcore Oy                    kingdom bleeds."
Systems. Networks. Security. -- George R.R. Martin: A Clash of Kings

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OK - I think we have converged on this, right?

Pekka Savola wrote:
> On Fri, 2 Feb 2007, Joe Touch wrote:
>>>    As a result, address filtering is not a local solution that can be=

>>>     deployed to protect communicating pairs, but rather relies on a
>>>     distributed infrastructure of trusted gateways filtering forged
>>>     traffic where it enters the network.  It is not feasible for loca=
l,
>>>     incremental deployment, and relies heavily on distributed
>>>     cooperation.  Although useful to reduce the load of spoofed traff=
ic,
>>>     it is insufficient to protect particular connections from attack
>>>     [29].
> ...
>>>  But a smaller delta to the current text might also be OK even if it =
is
>>>  less explicit, for example:
>>>
>>>     As a result, address filtering is not a local solution that can b=
e
>>>     deployed to protect communicating pairs, but rather relies on a
>>>     distributed infrastructure of trusted gateways filtering forged
>>>     traffic where it enters the network.  It is not feasible as a
>>>     general solution due to the lack universal deployment, but
>>>     may be applicable to connections among those inside the protected=

>>>     border in some scenarios.  Applying filtering also reduces
>>>     the load of spoofed traffic [29].
>>
>> I agree with the second sentence, which might be useful to include. As=

>> to the last sentence, this document focuses on endpoint mechanisms for=

>> defense, and doesn't address upstream load reduction.
>=20
> I don't care about the last sentence, and it could be removed or
> reworded.  It was just an attempt to rephrase the last sentence in the
> original (above) so that reference [29] is preserved.
>=20

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


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On 2007-2-3, at 8:41, ext Joe Touch wrote:
> Pekka Savola wrote:
>> (As I already said earlier, I believe this is a document that  
>> should be IETF Last Called even if doing so is not required by  
>> RFC2026.)
>
> I'm not sure every document wouldn't fall into that category; I  
> disagree that this particular document should be treated uniquely.

IETF Last Calls for non-standards-track docs are common, especially  
when a document should be brought to the attention to the wider  
community due to its importance or impact. (For example, we IETF-last- 
called QuickStart and the fragmentation-harmful documents in the  
recent past.)

I agree with Pekka that antispoof is sufficiently important that the  
wider community should be notified of its availability for review.

Lars



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From tcpm-bounces@ietf.org Mon Feb 05 09:26:45 2007
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From: Mark Allman <mallman@icir.org>
Subject: Re: [tcpm] DoS attack from misbehaving receivers 
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I have thought abou this attack and had some spirited discussion with
Rob on the topic.  I am not greatly worried about it myself.

  + It is pretty easy to detect this attack (in fact there is a footnote
    in the paper that says the author's experiments were detected by
    their network provider!).  It is fairly easy to see that more data
    is being ACKed than is being actually received.

  + Slammer was a one packet UDP fire and forget situation.  It sourced
    traffic as fast as the attached link could support---no control of
    any kind.  (Other worms have had this property, as well.)  We did
    not see a melting core.  So, I am not quite sure I am worried about
    a botnet of optack-ing boxes coaxing something to happen that has
    not already happened.

  + If we think this is a problem that needs a solution, we should think
    about how to do so without hacking things like changing the sending
    order.  (E.g., a generalized nonce (a la Savage) or something.

My two bits ...

allman




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On Wed, Jan 17, 2007 at 05:24:41PM -0800, Ted Faber wrote:
> 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.=20

After looking over the discussion, the chairs feel that (assuming
Pekka's issues have been suitably addressed) this document has passed
WGLC.  We'll plan to make the revisions that address Pekka's comments
and then advance this to the IESG.

Given Pekka's interest and Lars's support for an IETF last call, I
believe we'll request one.

Thanks again for your input.

--=20
Ted Faber
http://www.isi.edu/~faber           PGP: http://www.isi.edu/~faber/pubkeys.=
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Ted Faber wrote:
> On Wed, Jan 17, 2007 at 05:24:41PM -0800, Ted Faber wrote:
>> 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.=20
>=20
> After looking over the discussion, the chairs feel that (assuming
> Pekka's issues have been suitably addressed) this document has passed
> WGLC.  We'll plan to make the revisions that address Pekka's comments

Also Afred Hoenes, who pointed out a number of editorial errors I can
catch at this point, and a nominal tech issue as well. I'll post a
summary of changes to the list as soon as I get confirmation of
acceptance from them.

> and then advance this to the IESG.
>=20
> Given Pekka's interest and Lars's support for an IETF last call, I
> believe we'll request one.
>=20
> Thanks again for your input.
>=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 Mon Feb 05 16:42:15 2007
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From: Rob Sherwood <capveg@cs.umd.edu>
To: Mark Allman <mallman@icir.org>
Subject: Re: [tcpm] DoS attack from misbehaving receivers
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Hi Mark,

On Mon, Feb 05, 2007 at 09:24:25AM -0500, Mark Allman wrote:
> I have thought abou this attack and had some spirited discussion with
> Rob on the topic.  I am not greatly worried about it myself.
> 
>   + It is pretty easy to detect this attack (in fact there is a footnote
>     in the paper that says the author's experiments were detected by
>     their network provider!).  It is fairly easy to see that more data
>     is being ACKed than is being actually received.

If I recall correctly, the majority of the previous spirited discussions
surrounded this point :-)  To summarize my end of that discussion,
the attacker's ACK rate creates a trade-off between the detectability
of the attack versus the attack's amplification factor.  The paper focuses
on the theoretic maximum attack rate, i.e., sending one ACK per window,
without concern of detection.

I believe that an intelligent attacker can construct ACKs with reduced
amplification that exactly mimic the ACK stream of a valid connection,
making this attack impossible to detect in general given only the sender's
local information.  However, even with the amplification rate reduced to 
avoid detection, the amount of traffic generated is still significant.

For example from RFC2581: Sec 4.2, the attacker could generate ACKs for
every other segment, and space each ACK 100ms apart, in effect emulating
a slow connection (2 segments/100ms=~ 3000bytes/100ms=~ 29KB/s).  It is
not clear to me that there exists any method for the sender to locally,
passively differentiate this malicious connection from a completely
valid one because the ACK streams are identical.

While a single connection at this speed is uninteresting for DDoS, the
attacker could repeat this, in parallel, across many hosts, resulting
with a large aggregate amount of traffic, only limited by the attacker's
ability to produce ACKs.  Sending ACKs for every other segment produces
an amplification factor of 3000/40 or 75x the attacker's bandwidth,
implying that a single attacker on a 100Mb/s link (for example, at a
University) could in theory generate 7500Mb/s of traffic, which is more
than a OC-96 link (4976.64 Mb/s) and most of the capacity of an OC-192
link (9953.28 Mbit/s ).  So, this is a variant on the attack that is both
difficult to detect and has potential to be dangerous.

Further, the footnote is somewhat misleading.  The point of that
particular experiment was to maximize the amount of traffic induced from
a single host, not to avoid detection.  Had the experiment tried to avoid
detection, for example, by using the ACK rates described above, it is
highly unlikely the attack would have been detected.  Operators simply
cannot have alerts generated for every ~30KB/s stream on their network.


>   + Slammer was a one packet UDP fire and forget situation.  It sourced
>     traffic as fast as the attached link could support---no control of
>     any kind.  (Other worms have had this property, as well.)  We did
>     not see a melting core.  So, I am not quite sure I am worried about
>     a botnet of optack-ing boxes coaxing something to happen that has
>     not already happened.

But the set of hosts susceptible to the optack attack (i.e., everything
that implements TCP) is orders of magnitude larger then the set of hosts
susceptible to the Slammer attack (i.e., everything that ran Microsoft's
SQL server).  I assume someone will correct me if I'm wrong, but I do not
believe we have seen a worm attack that caused even the majority of access
links to saturate, which is exactly what I contend optack+botnets can do.

>   + If we think this is a problem that needs a solution, we should think
>     about how to do so without hacking things like changing the sending
>     order.  (E.g., a generalized nonce (a la Savage) or something.

I agree that the generalized nonce solution (particularly, the cumulative
nonce solution proposed by Savage et al.) is a more elegant solution.
However, it requires both sender and receiver code to change, limiting
its incremental deployability.  If the deployability issues could be
addressed, this is clearly the right way to go, IMHO.

- Rob
.

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To: Rob Sherwood <capveg@cs.umd.edu>
From: Mark Allman <mallman@icir.org>
Subject: Re: [tcpm] DoS attack from misbehaving receivers 
In-Reply-To: <20070205214141.GA11207@loompa.cs.umd.edu> 
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Rob-

> >   + It is pretty easy to detect this attack (in fact there is a footnote
> >     in the paper that says the author's experiments were detected by
> >     their network provider!).  It is fairly easy to see that more data
> >     is being ACKed than is being actually received.
>
> [...]
> 
> I believe that an intelligent attacker can construct ACKs with reduced
> amplification that exactly mimic the ACK stream of a valid connection,
> making this attack impossible to detect in general given only the
> sender's local information.  However, even with the amplification rate
> reduced to avoid detection, the amount of traffic generated is still
> significant.

Maybe I was not clear enough above.  I think I agree with you that
detection on the sender-side is hard---although, a sender that is
continuously pounding is actually quite noisy and getting only ACKs that
acknowledge large amounts of sequence space is suggestive.

What I meant was that at the receiver side you can detect this attack
pretty readily.  If X bytes are received and 10X bytes are ACKed
(regardless of how many ACKs are used or how they are spaced) then
something is wrong.  

Basically, in terms of sequence space you cannot mimic the flow normal
flow of ACKs and get any amplification.  If you want to amplify then you
have to change things and those changes can be observed.

> Further, the footnote is somewhat misleading.  The point of that
> particular experiment was to maximize the amount of traffic induced
> from a single host, not to avoid detection.  Had the experiment tried
> to avoid detection, for example, by using the ACK rates described
> above, it is highly unlikely the attack would have been detected.

I understand.  I found the footnote to be illustrative in that
conceivably nobody was looking for your 'attack' traffic and yet they
found it and took action anyway.  To me this says that if you want to
amplify and maximize things then people will find you.

> Operators simply cannot have alerts generated for every ~30KB/s stream
> on their network.

I agree.  However, I can't believe you are actually suggesting using
optacks to get 30KB/sec flows.  You don't need optack at that point.
Further, for some detection methods (i.e., more being ACKed than
received) you don't need an alert---you just need to send a RST to the
server and kill the connection because it isn't doing any useful work.

> >   + Slammer was a one packet UDP fire and forget situation.  It sourced
> >     traffic as fast as the attached link could support---no control of
> >     any kind.  (Other worms have had this property, as well.)  We did
> >     not see a melting core.  So, I am not quite sure I am worried about
> >     a botnet of optack-ing boxes coaxing something to happen that has
> >     not already happened.
> 
> But the set of hosts susceptible to the optack attack (i.e., everything
> that implements TCP) is orders of magnitude larger then the set of hosts
> susceptible to the Slammer attack (i.e., everything that ran Microsoft's
> SQL server).

Three things ...

  + Optack is a secondary attack.  That is, a host will only do
    optimistic ACKing once it has been compromised in some other
    fashion.  So, saying that Slammer is somehow a bad example because
    it is "small" is not quite right, IMO.

  + I understand that one compromised machine can optack against
    somewhat arbitrary servers.  (Although, note that you need to find
    places to slurp from that actually have a reasonable amount of data
    such that they could in fact impose a load on the network.)  But,
    you are still going to be somewhat limited by the number of
    compromised hosts you can bring to bear on the DDoS.

  + My point in bringing up Slammer is not that it is precisely the same
    as what you are suggesting.  My point is that this was a case where
    75K hosts were pounding the network as hard as they possibly could
    and the core did not melt and there was no congestion collapse.  I
    understand that you could theoretically bring more firepower to bear
    on the network, but I am not sure that in practical terms you could
    ever really do it.

> I agree that the generalized nonce solution (particularly, the
> cumulative nonce solution proposed by Savage et al.) is a more elegant
> solution.  However, it requires both sender and receiver code to
> change, limiting its incremental deployability.  If the deployability
> issues could be addressed, this is clearly the right way to go, IMHO.

What is the "deployability" issue?  That it cannot happen as quickly as
a sender-only hack?  This all comes down to (1) do we think this is a
problem that really needs to be addressed and, if so then (2) do we
think this needs to be addressed immediately or would a cleaner approach
that took a bit longer to engineer and deploy be fine?  I am not sure I
even think this is a huge problem we are facing and so I am not in favor
of short term hacks when we know how to do better.  I understand that we
disagree on this point (which is fine).

allman




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From: Rob Sherwood <capveg@cs.umd.edu>
To: Mark Allman <mallman@icir.org>
Subject: Re: [tcpm] DoS attack from misbehaving receivers
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On Mon, Feb 05, 2007 at 09:14:46PM -0500, Mark Allman wrote:
> Maybe I was not clear enough above.  I think I agree with you that
> detection on the sender-side is hard---although, a sender that is
> continuously pounding is actually quite noisy and getting only ACKs that
> acknowledge large amounts of sequence space is suggestive.

My confusion - I thought you were referring to detecting this on the
sender's side.  I agree that detecting the attack on the receiving
end is more straight forward, but it's not clear to me that relying
on receiver-side network operators to solve this problem is always
practical/desirable.  While this is a more philosophical argument,
it is a well established one in the DDoS literature: if your network
is under attack, you want to solve the problem locally, not depend on
operators at the source of the attack to solve it for you.  I consider
the issues with IP spoofing and the lack of pervasive egress filtering
a reasonable data point of why source-side solutions are not practical.
However, if this group feels that source-based prevention techniques
are sufficient, then I guess I should stop bugging you :-)

> I agree.  However, I can't believe you are actually suggesting using
> optacks to get 30KB/sec flows.  You don't need optack at that point.

I was suggesting that you need optack to get *many* 30KB/s flows, where
the total aggregate traffic is larger than possible without optack.  

> Three things ...
> 
>   + Optack is a secondary attack.  That is, a host will only do
>     optimistic ACKing once it has been compromised in some other
>     fashion.  So, saying that Slammer is somehow a bad example because
>     it is "small" is not quite right, IMO.
>
>   + I understand that one compromised machine can optack against
>     somewhat arbitrary servers.  (Although, note that you need to find
>     places to slurp from that actually have a reasonable amount of data
>     such that they could in fact impose a load on the network.)  But,
>     you are still going to be somewhat limited by the number of
>     compromised hosts you can bring to bear on the DDoS.

I think I've been unclear.  So if an attacker compromises K machines,
and uses them to optack attack N hosts, then you are saying that the
attack is limited because an attacker cannot make K big enough?  I think
I'm missing something, but my assumption was that this the attacker
would be launching this from a botnet, which there are studies that
show the existence of botnets on the order of 10K+ hosts and larger.
Having a K in the 10K range with any sort of amplification factor would
seem to be a significant threat to me.

Additionally, I'm assuming the reason this issue was brought back up (and
to this list) was that Michal Zalewski recently found a vulnerability in
deployed versions of Apache and IIS that allows for a malicious client
to request arbitrary data from a web server.  While this is not terribly
interesting in general, it means that every web server, not just those
with large files, are now potential optack targets.

http://www.securityfocus.com/archive/1/455833/100/0/threaded

>   + My point in bringing up Slammer is not that it is precisely the same
>     as what you are suggesting.  My point is that this was a case where
>     75K hosts were pounding the network as hard as they possibly could
>     and the core did not melt and there was no congestion collapse.  I
>     understand that you could theoretically bring more firepower to bear
>     on the network, but I am not sure that in practical terms you could
>     ever really do it.

Well, the short story is that I'm not 100% certain that the attack
is practical either.  My current research involves trying to decide
the attack's practicality  without actually running it.  This includes
some network topology mapping to evaluate the Internet's vulnerability
to optack and non-optack bandwidth flooding attacks, and simulations to
ensure that the attack is not self-limiting.  However, I think it is safe
to say that even if it is not practical to cause collapse in the core,
there is still danger to access links such that it is still worth while
spending time discussing possible solutions.

> What is the "deployability" issue?  That it cannot happen as quickly as
> a sender-only hack?  This all comes down to (1) do we think this is a
> problem that really needs to be addressed and, if so then (2) do we
> think this needs to be addressed immediately or would a cleaner approach
> that took a bit longer to engineer and deploy be fine?  I am not sure I
> even think this is a huge problem we are facing and so I am not in favor
> of short term hacks when we know how to do better.  I understand that we
> disagree on this point (which is fine).

So I have backed off of my original stance w.r.t (2) since our previous
conversation about this: the paper has been out for over a year now and
people are not exploiting this attack in the wild, so clearly there is
time to engineer a better solution if one is available.

That said, I maintain that any proper solution should only modify the
sender, where existing cleaner solutions modify both sender and receiver.
Because the change affects security, any New Solution should not be
optional or negotiated at connection time - both ends of the connection
need to support it, else the attack vector persists and the solution
accomplishes nothing.  It does no good to have a solution where the client
can just pretend to not support it.  If we require that all senders and
receivers implement New Solution, then it creates interoperability issues
with TCP implementations that do and do not yet support New Solution.
This is what I mean by the deployability issue.

- Rob
.

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From: Mark Allman <mallman@icir.org>
Subject: Re: [tcpm] DoS attack from misbehaving receivers 
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> My confusion - I thought you were referring to detecting this on the
> sender's side.  I agree that detecting the attack on the receiving end
> is more straight forward, but it's not clear to me that relying on
> receiver-side network operators to solve this problem is always
> practical/desirable.  While this is a more philosophical argument, it
> is a well established one in the DDoS literature: if your network is
> under attack, you want to solve the problem locally, not depend on
> operators at the source of the attack to solve it for you.  I consider
> the issues with IP spoofing and the lack of pervasive egress filtering
> a reasonable data point of why source-side solutions are not
> practical.  However, if this group feels that source-based prevention
> techniques are sufficient, then I guess I should stop bugging you :-)

But, your line of thinking here is that it is the *network* that is
under attack, right?  I.e., the compromised machine (the sink) and the
machine it is coaxing into a higher-than-usual sending rate are merely
pawns and there may be no big effects on either of these hosts or their
networks, right?  So, I am not sure the experience you cite above is
germane.  

Also, in thinking about this more I think you could detect *something
weird* on the sender side of the attack (i.e., the side receiving the
optacks).  One could think in terms of the symmetry work that Kreibich,
et.al. presented at hotnets a couple years ago.  The sender-side would
certainly see a much larger ACK ratio than one ACK / two data packets,
right?  (If not then there is no amplification.)  This is not as good of
a test as the receiver-side test of ACKing more data than arrives in
that it is not iron clad.

> Additionally, I'm assuming the reason this issue was brought back up
> (and to this list) was that Michal Zalewski recently found a
> vulnerability in deployed versions of Apache and IIS that allows for a
> malicious client to request arbitrary data from a web server.  While
> this is not terribly interesting in general, it means that every web
> server, not just those with large files, are now potential optack
> targets.
> 
> http://www.securityfocus.com/archive/1/455833/100/0/threaded

Oh, ugh.  (With or without optack.)

> Well, the short story is that I'm not 100% certain that the attack is
> practical either.  My current research involves trying to decide the
> attack's practicality without actually running it.  This includes some
> network topology mapping to evaluate the Internet's vulnerability to
> optack and non-optack bandwidth flooding attacks, and simulations to
> ensure that the attack is not self-limiting.  However, I think it is
> safe to say that even if it is not practical to cause collapse in the
> core, there is still danger to access links such that it is still
> worth while spending time discussing possible solutions.

I look forward to seeing this work.

> So I have backed off of my original stance w.r.t (2) since our
> previous conversation about this: the paper has been out for over a
> year now and people are not exploiting this attack in the wild, so
> clearly there is time to engineer a better solution if one is
> available.

(And, the attack was made public by others before your paper in an I-D.
Even presented at a TCPM meeting, I believe.  (Or, maybe tsvwg.))

> That said, I maintain that any proper solution should only modify the
> sender, where existing cleaner solutions modify both sender and
> receiver.  Because the change affects security, any New Solution
> should not be optional or negotiated at connection time - both ends of
> the connection need to support it, else the attack vector persists and
> the solution accomplishes nothing.  It does no good to have a solution
> where the client can just pretend to not support it.  If we require
> that all senders and receivers implement New Solution, then it creates
> interoperability issues with TCP implementations that do and do not
> yet support New Solution.  This is what I mean by the deployability
> issue.

I understand this take.  Certainly there is logic in here.  But, let me
come at it from a different angle.  Security is all bout policy.  So,
maybe big web servers that don't want to be attacked just rate limit
connections unless they use the new mechanism.  Say, they allow some
number of bytes to be sent without restriction, but if something big is
being transmitted (i.e., we're in the heavy tail) then some client is
not going to get good performance unless they turn on New Solution.  If
they don't then they will get suboptimal performance.  Of course, if
we're looking at things in this detail a policy could also be to limit
bandwidth when connections are sending lots of data, seeing no loss and
seeing big stretch ACKs and we don't need a mechanism at all.

The other thing that is still rolling around my head is that this is
claimed (rightly, IMO) as an attack on the *network*.  Further, you are
saying that a jillion tiny flows can make up the attack.  Fine.  So, it
is not clear to me why any end host has much incentive to do something
to help this problem much.  It is not going to be an issue at the edges
of the networks, only when things aggregate up.  Of course, then also
everyone has an incentive to keep the shared part of the network working
well---which would then argue that a scheme that involved both endpoints
might work out OK.

I am still not convinced that we need to mess with the protocols at all
here, however.

allman




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Subject: [tcpm] WGLC draft-ietf-behave-tcp-04 completed
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The WGLC of draft-ietf-behave-tcp-04 has completed, with no comments
received.  The document will be sent to IESG for their review.

PROTO writeup is below.

-d

-----

PROTO (draft-ietf-proto-wgchair-doc-shepherding-09) writeup for:

  "NAT Behavioral Requirements for TCP",
  http://www.ietf.org/internet-drafts/draft-ietf-behave-tcp-04.txt



   (1.a)  Who is the Document Shepherd for this document?  Has the
          Document Shepherd personally reviewed this version of the
          document and, in particular, does he or she believe this
          version is ready for forwarding to the IESG for publication?

Dan Wing

The document has been reviewed and is ready for forwarding to IESG for 
publication.


   (1.b)  Has the document had adequate review both from key WG members
          and from key non-WG members?  Does the Document Shepherd have
          any concerns about the depth or breadth of the reviews that
          have been performed?

During its development, the document has received input from members
of the TCPM working group (especially Joe Touch and Fernando Gont), and 
that input was integrated into this document.

The document had a two-week WGLC in both TCPM and BEHAVE working groups.  

There were no comments during this last call.


The Document Shepherd has no concerns about the depth or breadth of
the reviews.


   (1.c)  Does the Document Shepherd have concerns that the document
          needs more review from a particular or broader perspective,
          e.g., security, operational complexity, someone familiar with
          AAA, internationalization or XML?

Additional review is not necessary.


   (1.d)  Does the Document Shepherd have any specific concerns or
          issues with this document that the Responsible Area Director
          and/or the IESG should be aware of?  For example, perhaps he
          or she is uncomfortable with certain parts of the document, or
          has concerns whether there really is a need for it.  In any
          event, if the WG has discussed those issues and has indicated
          that it still wishes to advance the document, detail those
          concerns here.  Has an IPR disclosure related to this document
          been filed? If so, please include a reference to the disclosure 
          and summarize the WG discussion and conclusion on this issue.

There have been a little concern that the TCP document normatively 
references the UDP document (draft-ietf-behave-nat-udp, soon to be 
published as RFC4787), because this requires reading both documents.
WG consensus was to normatively reference UDP, as is done in the
document.

There is no IPR on this document.


   (1.e)  How solid is the WG consensus behind this document?  Does it
          represent the strong concurrence of a few individuals, with
          others being silent, or does the WG as a whole understand and
          agree with it?

The document has strong WG consensus.


   (1.f)  Has anyone threatened an appeal or otherwise indicated extreme
          discontent?  If so, please summarise the areas of conflict in
          separate email messages to the Responsible Area Director.  (It
          should be in a separate email because this questionnaire is
          entered into the ID Tracker.)

None indicated.


   (1.g)  Has the Document Shepherd personally verified that the
          document satisfies all ID nits?  (See
          http://www.ietf.org/ID-Checklist.html and
          http://tools.ietf.org/tools/idnits/).  Boilerplate checks are
          not enough; this check needs to be thorough.  Has the document
          met all formal review criteria it needs to, such as the MIB
          Doctor, media type and URI type reviews?

The document hsa pre-Feb-2007 copyright, per idnits 1.124:

  - This document has ISOC Copyright according to RFC 3978, instead of the
    newer IETF Trust Copyright according to RFC 4748.  You should consider
    updating it; the new Copyright statement will be required from February
    1st, 2007
  - This document has an original RFC 3978 Section 5.5 Disclaimer, instead
of
    the newer disclaimer which includes the IETF Trust according to RFC
4748. 
    You should consider updating it; the new disclaimer will be required
from
    February 1st, 2007


   (1.h)  Has the document split its references into normative and
          informative?  Are there normative references to documents that
          are not ready for advancement or are otherwise in an unclear
          state?  If such normative references exist, what is the
          strategy for their completion?  Are there normative references
          that are downward references, as described in [RFC3967]?  If
          so, list these downward references to support the Area
          Director in the Last Call procedure for them [RFC3967].


Yes, the document has split its references.  

There are two I-Ds cited as normative:  

  * draft-ietf-behave-nat-icmp, which is not yet ready for 
    advancement (it has not been WGLC'd).  It is expected to be
    WGLC'd later this year.
  * draft-ietf-behave-nat-udp is in the RFC Editor's queue.

There are no downward references.


   (1.i)  Has the Document Shepherd verified that the document IANA
          consideration section exists and is consistent with the body
          of the document?  If the document specifies protocol
          extensions, are reservations requested in appropriate IANA
          registries?  Are the IANA registries clearly identified?  If
          the document creates a new registry, does it define the
          proposed initial contents of the registry and an allocation
          procedure for future registrations?  Does it suggest a
          reasonable name for the new registry?  See
          [RFC2434.  If the document describes an Expert Review process 
          has Shepherd conferred with the Responsible Area Director so that 
          the IESG can appoint the needed Expert during the IESG Evaluation?

There are no IANA considerations for this specification; the document does
not describe an Expert Review process.


   (1.j)  Has the Document Shepherd verified that sections of the
          document that are written in a formal language, such as XML
          code, BNF rules, MIB definitions, etc., validate correctly in
          an automated checker?

This document does not contain any such formal language.


   (1.k)  The IESG approval announcement includes a Document
          Announcement Write-Up.  Please provide such a Document
          Announcement Write-Up?  Recent examples can be found in the
          "Action" announcements for approved documents.  The approval
          announcement contains the following sections:

          Technical Summary
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.


          Working Group Summary
This document was a product of the BEHAVE working group.


          Document Quality
This document describes recommended practices for NATs.  Most 
existing NATs already conform to these requirements.


          Personnel
Dan Wing is the Document Shepherd, and Magnus Westerlund is
the Responsible Area Director.


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

I received this feedback from Alfred Hoenes. I will most my answers soon.


>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)
>
>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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Folks,

We have published a revision of the port randomization draft 
(draft-larsen-tsvwg-port-randomization). This version addresses 
feedback from Alfred Hoenes and Carlos Pignataro, and comments from 
FreeBSD's Mike Silbersack. The draft is targetted at tsvwg because 
the same stuff can be applied to other protocols. But most (all?) of 
the work on this has been done mainly for TCP.

The draft is available at 
http://www.gont.com.ar/drafts/port-randomization/draft-larsen-tsvwg-port-randomization-01.txt 
, and will soon be available at the usual places.

Any feedback will be more than welcome.

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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Folks-

We have just submitted a new rev of 2581bis.  We fixed some minor things
that folks flagged.  In addition, we made two substantive changes, which
I am putting into two different emails.  

The draft should pop out of the usual places soon, but until then you
can slurp a copy from:

    http://www.icir.org/mallman/papers/draft-ietf-tcpm-rfc2581bis-02.txt

The first issue we discussed a while back (in Montreal) and I believe
the room had decent consensus on this, but it took a while to roll it
into the draft and therefore verify it here on the list.

The change...  On the first retransmission of a segment with sequence
number X the ssthresh is set to half the FlightSize as it always has
been specified.  However, on subsequent retransmissions the same segment
ssthresh is held constant.  RFC 2581 was ambiguous on this point.
However, the intent was that ssthresh be halved on each retransmit.
However, the WG seems to have consensus that this is not really required
and the upside is that we do not doom connections to linear growth hell
for ephemeral network conditions.

Why I think this is OK is that if it is a problem that we held ssthresh
constant and did not drop it then we will quickly learn of this by
taking more loss and therefore collapsing ssthresh further.

If folks feel that we have taken the wrong approach in the document,
please yell.

Thanks,
allman




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Folks-

We have just submitted a new rev of 2581bis.  We fixed some minor things
that folks flagged.  In addition, we made two substantive changes, which
I am putting into two different emails.  

The draft should pop out of the usual places soon, but until then you
can slurp a copy from:

    http://www.icir.org/mallman/papers/draft-ietf-tcpm-rfc2581bis-02.txt

The first issue we discussed a while back (in Montreal) and I believe
the room had decent consensus on this, but it took a while to roll it
into the draft and therefore verify it here on the list.

The change...  On the first retransmission of a segment with sequence
number X the ssthresh is set to half the FlightSize as it always has
been specified.  However, on subsequent retransmissions the same segment
ssthresh is held constant.  RFC 2581 was ambiguous on this point.
However, the intent was that ssthresh be halved on each retransmit.
However, the WG seems to have consensus that this is not really required
and the upside is that we do not doom connections to linear growth hell
for ephemeral network conditions.

Why I think this is OK is that if it is a problem that we held ssthresh
constant and did not drop it then we will quickly learn of this by
taking more loss and therefore collapsing ssthresh further.

If folks feel that we have taken the wrong approach in the document,
please yell.

Thanks,
allman




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Subject: [tcpm] rfc2581bis [2]: limiting cwnd inflation
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Reminder, the draft should pop out of the usual places soon, but until
then you can slurp a copy from:

    http://www.icir.org/mallman/papers/draft-ietf-tcpm-rfc2581bis-02.txt

The second issue ...

We added a note to step 4 of the fast retransmit/recovery procedure:

    4.  For each additional duplicate ACK received (after the third),
        cwnd MUST be incremented by SMSS.  This artificially inflates
        the congestion window in order to reflect the additional segment
        that has left the network.
    
        Note: [SCWA99] discusses a receiver-based attack whereby many
        bogus duplicate ACKs are sent to the data sender in order to
        artificially inflate cwnd and cause a higher than appropriate
        sending rate to be used.  A TCP MAY therefore limit the number
        of times cwnd is artificially inflated during loss recovery
        to the number of outstanding segments (or, an approximation
        thereof). 

That is, we explicitly allow a TCP to limit the number of dupacks that
are acceptable during fast recovery to prevent an attack such as
discussed in [SCWA99].

Note that (1) this is more conservative than the algorithm in 2581 and
so implicitly allowed already (that is, we are just calling attention to
this) and (2) we are not trying to mandate this behavior, just noting
that it might be useful.

Comments would be appreciated.

Thanks,
allman




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Subject: [tcpm] rfc2581bis [2]: limiting cwnd inflation
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Reminder, the draft should pop out of the usual places soon, but until
then you can slurp a copy from:

    http://www.icir.org/mallman/papers/draft-ietf-tcpm-rfc2581bis-02.txt

The second issue ...

We added a note to step 4 of the fast retransmit/recovery procedure:

    4.  For each additional duplicate ACK received (after the third),
        cwnd MUST be incremented by SMSS.  This artificially inflates
        the congestion window in order to reflect the additional segment
        that has left the network.
    
        Note: [SCWA99] discusses a receiver-based attack whereby many
        bogus duplicate ACKs are sent to the data sender in order to
        artificially inflate cwnd and cause a higher than appropriate
        sending rate to be used.  A TCP MAY therefore limit the number
        of times cwnd is artificially inflated during loss recovery
        to the number of outstanding segments (or, an approximation
        thereof). 

That is, we explicitly allow a TCP to limit the number of dupacks that
are acceptable during fast recovery to prevent an attack such as
discussed in [SCWA99].

Note that (1) this is more conservative than the algorithm in 2581 and
so implicitly allowed already (that is, we are just calling attention to
this) and (2) we are not trying to mandate this behavior, just noting
that it might be useful.

Comments would be appreciated.

Thanks,
allman




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Message-ID: <45D20253.70706@cisco.com>
Date: Tue, 13 Feb 2007 10:24:19 -0800
From: Mahesh Jethanandani <mahesh@cisco.com>
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A new I-D has been posted on the IETF web site.

http://www.ietf.org/internet-drafts/draft-mahesh-persist-timeout-00.txt
"TCP Maintenance and Minor Extensions", Mahesh Jethanandani, Murali Bashyam, 9-Feb-07, <draft-mahesh-persist-timeout-00.txt> 

Comments are welcome.
-- 

*Mahesh Jethanandani*
*Technical Leader*
**Corporate Development*
*
mahesh@cisco.com <mailto:mahesh@cisco.com>
Phone :*+1 408 527-8230*
Fax :*+1 408 527-6351*

	

**
170 West Tasman Drive
San Jose, CA 95070
United States
www.cisco.com <http://www.cisco.com>

	 

This e-mail may contain confidential and privileged material for the 
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A new I-D has been posted on the IETF web site.<br>
<pre wrap=""><a class="moz-txt-link-freetext"
 href="http://www.ietf.org/internet-drafts/draft-mahesh-persist-timeout-00.txt">http://www.ietf.org/internet-drafts/draft-mahesh-persist-timeout-00.txt</a>
"TCP Maintenance and Minor Extensions", Mahesh Jethanandani, Murali Bashyam, 9-Feb-07, &lt;draft-mahesh-persist-timeout-00.txt&gt; </pre>
Comments are welcome.<br>
<div class="moz-signature">-- <br>
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 style="background: transparent url(http://www.cisco.com/global/EMEA/brand/signature/nordics/bg3.jpg) no-repeat scroll center top; -moz-background-clip: -moz-initial; -moz-background-origin: -moz-initial; -moz-background-inline-policy: -moz-initial;"
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            <p
 style="font-family: Arial,Helvetica,sans-serif; font-size: 11px; font-weight: normal; color: rgb(102, 102, 102);"><strong>Mahesh
Jethanandani</strong><br>
            <strong>Technical Leader</strong><br>
            <strong><strong>Corporate Development</strong><br>
            </strong><br>
            <a style="color: rgb(102, 102, 102);"
 href="mailto:mahesh@cisco.com">mahesh@cisco.com</a><br>
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 nowrap="nowrap" valign="top">
            <p
 style="font-family: Arial,Helvetica,sans-serif; font-size: 11px; font-weight: normal; color: rgb(102, 102, 102);"><strong></strong><br>
170 West Tasman Drive<br>
San Jose, CA 95070<br>
United States<br>
            <a style="color: rgb(102, 102, 102);"
 href="http://www.cisco.com">www.cisco.com</a></p>
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            <td width="200">&nbsp;</td>
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            <td><img src="cid:part2.07080702.00040906@cisco.com"
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On Tue, Feb 13, 2007 at 10:24:19AM -0800, Mahesh Jethanandani wrote:
> A new I-D has been posted on the IETF web site.
> 
> http://www.ietf.org/internet-drafts/draft-mahesh-persist-timeout-00.txt
> "TCP Maintenance and Minor Extensions", Mahesh Jethanandani, Murali 
> Bashyam, 9-Feb-07, <draft-mahesh-persist-timeout-00.txt> 
> Comments are welcome.

It's good to see the description of the problem you observed along with
the solution, but it looks to me like this is more of an OS mechanism to
manage resources rather than a protocol-based mechanism.  Is this
correct?

-- 
Wesley M. Eddy
Verizon Federal Network Systems

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In-Reply-To: Your message of "Tue, 13 Feb 2007 10:24:19 PST."
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> A new I-D has been posted on the IETF web site.

> http://www.ietf.org/internet-drafts/draft-mahesh-persist-timeout-00.txt
> "TCP Maintenance and Minor Extensions", Mahesh Jethanandani, Murali Bashyam, 9

This idea seems quite closely related to TCP keepalives, though it
is addressing a different problem. Do you have any suggestions for
a sockets API to these persist timers?

The draft spends a lot of time showing that the application cannot
deal with the problem. I don't buy all the arguments in this section:
for example, an application can usually control the sockbuf sizes
and then either user non-blocking writes or timeouts to determine
if the application at the other end has gone away or not. I'm not
sure how often an application cares if the TCP at the other end is
still alive!

On the other hand, as pointed out, there are situations where there
is no longer any application (data has been handed off to TCP and
the application has exited/moved on). In these cases it could be
quite hard for an administrator to control the resource usage without
something like the timeouts that you describe. I know FreeBSD has
had to introduce some facilities for dealing with these sort of
resource issues (there is a per-process limit on how many sockbufs
can be allocated by a process and there is a "tcpdrop" utility to
force a TCP connection to be closed).

	David.

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From: MURALI BASHYAM <murali_bashyam@yahoo.com>
Subject: Re: [tcpm] New I-D
To: weddy@grc.nasa.gov, Mahesh Jethanandani <mahesh@cisco.com>
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It's a TCP protocol and connection state specific
solution to the infinite persist state issue. The
consequences of not hanndling this issue lead to a
resource starvation problem, along with other
problems, DDOS etc.

Murali

--- Wesley Eddy <weddy@grc.nasa.gov> wrote:

> On Tue, Feb 13, 2007 at 10:24:19AM -0800, Mahesh
> Jethanandani wrote:
> > A new I-D has been posted on the IETF web site.
> > 
> >
>
http://www.ietf.org/internet-drafts/draft-mahesh-persist-timeout-00.txt
> > "TCP Maintenance and Minor Extensions", Mahesh
> Jethanandani, Murali 
> > Bashyam, 9-Feb-07,
> <draft-mahesh-persist-timeout-00.txt> 
> > Comments are welcome.
> 
> It's good to see the description of the problem you
> observed along with
> the solution, but it looks to me like this is more
> of an OS mechanism to
> manage resources rather than a protocol-based
> mechanism.  Is this
> correct?
> 
> -- 
> Wesley M. Eddy
> Verizon Federal Network Systems
> 
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www1.ietf.org/mailman/listinfo/tcpm
> 



 
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Subject: RE: [tcpm] rfc2581bis [2]: limiting cwnd inflation
Date: Tue, 13 Feb 2007 21:13:13 -0800
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Thread-Topic: [tcpm] rfc2581bis [2]: limiting cwnd inflation
thread-index: AcdPgNS5TiMWicdXSxeAhZUi0PBTqAAcTtFg
From: "Anantha Ramaiah \(ananth\)" <ananth@cisco.com>
To: <mallman@icir.org>, <tcpm@ietf.org>
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I was having some private email discussions with Mark when the previous
version was released, basically talking about limiting the artifical
cwnd inflation during fast recovery. Now, the following text which is
added in the new version of this draft talks about the mis-behaving
receivers only. What I had in mind was about in general (whether or not
the receiver is mis-behaving, this applies). To illustrate this :-

Assume that flightsize =3D f during fast recovery. Assume f > 2 SMSS

When the  3rd duplicate Ack arrives,=20

ssthresh =3D max (FlightSize / 2, 2*SMSS)
Ssthresh =3D max ( f/2, 2*smss) =3D f/2.

Now cwnd =3D f/2 + 3*SMSS

For each duplicate ACK which arrives after that (assuming that the first
segment got dropped) *in theory* you could get 6 duplicate ACK's [This
is NOT a mis-behaving receiver] Since every out-of-order segment ought
to be ACKED immediatley and shouldn't be delayed.

So we would have incremented the cwnd by :-

Cwnd  =3D (f/2 + 3*SMSS) + 6SMSS > f.  This can happen since there seems
to be no explicit rule to cap the cwnd growth which may cause the higher
than appropriate sending rate to be used.  To make the above example
assuming f =3D 10 before fast recovery entered, we would be in a =
situation
where cwnd becomes 14!!, which is un-desirable.

So I would think the text needs to be more generic something like :-=20

"Note: Since, in theory, the cwnd can get artifically inflated to a
value greater than what it was before the fast recovery procedure was
entered, a TCP MAY limit the number of times cwnd is artificially
inflated during loss recovery to the number of outstanding segments (or,
an approximation thereof). Also this helps in dealing with
receiver-based attack wherby many bogus duplicate ACKS are sent to the
data sender in order to artifically inflate the cwnd and cause higher
than appropriate sending rate to be used."

Mark suggested that I send this to the wider audience to see if there is
some general consensus on this.

-Anantha
> -----Original Message-----
> From: Mark Allman [mailto:mallman@icir.org]=20
> Sent: Tuesday, February 13, 2007 7:04 AM
> To: tcpm@ietf.org
> Subject: [tcpm] rfc2581bis [2]: limiting cwnd inflation
>=20
> =20
> Reminder, the draft should pop out of the usual places soon,=20
> but until then you can slurp a copy from:
>=20
>    =20
> http://www.icir.org/mallman/papers/draft-ietf-tcpm-rfc2581bis-02.txt
>=20
> The second issue ...
>=20
> We added a note to step 4 of the fast retransmit/recovery procedure:
>=20
>     4.  For each additional duplicate ACK received (after the third),
>         cwnd MUST be incremented by SMSS.  This artificially inflates
>         the congestion window in order to reflect the=20
> additional segment
>         that has left the network.
>    =20
>         Note: [SCWA99] discusses a receiver-based attack whereby many
>         bogus duplicate ACKs are sent to the data sender in order to
>         artificially inflate cwnd and cause a higher than appropriate
>         sending rate to be used.  A TCP MAY therefore limit the number
>         of times cwnd is artificially inflated during loss recovery
>         to the number of outstanding segments (or, an approximation
>         thereof).=20
>=20
> That is, we explicitly allow a TCP to limit the number of=20
> dupacks that are acceptable during fast recovery to prevent=20
> an attack such as discussed in [SCWA99].
>=20
> Note that (1) this is more conservative than the algorithm in=20
> 2581 and so implicitly allowed already (that is, we are just=20
> calling attention to
> this) and (2) we are not trying to mandate this behavior,=20
> just noting that it might be useful.
>=20
> Comments would be appreciated.
>=20
> Thanks,
> allman
>=20
>=20
>=20
>=20

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To: "Anantha Ramaiah (ananth)" <ananth@cisco.com>
From: Mark Allman <mallman@icir.org>
Subject: Re: [tcpm] rfc2581bis [2]: limiting cwnd inflation 
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> Cwnd  = (f/2 + 3*SMSS) + 6SMSS > f.  This can happen since there seems
> to be no explicit rule to cap the cwnd growth which may cause the higher
> than appropriate sending rate to be used.  To make the above example
> assuming f = 10 before fast recovery entered, we would be in a situation
> where cwnd becomes 14!!, which is un-desirable.

It is not.  After out iteration earlier today I went through this in
more detail in my head.  You want cwnd to grow past where it was before
recovery.  That is how you're able to strobe new segments out.  Simple
example:

FlightSize=40
lose 1 segment; expect 39 dupacks (say none are lost)
after the 3rd dupack:
  cwnd = (FlightSize / 2) + 3 = 23
  ssthresh = FlightSize / 2 = 20
  retransmit
receiver 36 more dupacks, on each:
  cwnd += 1
  if FlightSize < cwnd:
    transmit
    FlightSize += 1
receive an ACK for everything
  cwnd = ssthresh

So, basically before the rexmt is ACKed cwnd will be 59 segments---or,
19 more than it was before the loss happened.  But, this increase is
termed *artificial*.  We did not put 59 segments into the network during
the last RTT, we put 19 segments into the network (or, half of the
FlightSize when the loss occured)---so, we did in fact halve our rate
even though the cwnd is larger.  Before continuing the cwnd collapses
back to what it ought to be (20 segments in this case).

What the new text is attempting to do is to allow stacks to protect
against malicious receivers that strobe many more dupacks into the
stream to boost the sending rate.  Regardless of the value of cwnd, I
don't think there is a case whereby you'll be actually putting an
inappropriate number of segments into the network when things are
working properly.

allman




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Subject: [tcpm] Some comments on 2581bis
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Mark,

Some comments on the draft:

** RFC 2581 & SACK
Page 3 says:
         Alternatively, a TCP that utilizes selective acknowledgments
         [RFC2018,RFC2883] can determine an incoming ACK is a "duplicate"
         if the ACK contains previously unknown SACK information.

And page 7 says:
Note that a sender using SACK [RFC2018]
         MUST NOT send new data unless the incoming duplicate
         acknowledgment contains new SACK information.

IMHO, if you keep the "MUST NOT" in page 7, you'd probably 
s/can/MUST/ in page 3. Also, maybe it would be worth to clarify that 
the SACK information should be "in window"?


** Limits on dupacks
Would it make sense to limit the number of accepted dupacks to 
Max_dupacks = (Flight_size_fr - 1) / SMSS ?
where Flight_size_fr is the Flight_Size at the point fast retransmit 
is triggered.

Thanks,

-- 
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e-mail: fernando@gont.com.ar || fgont@acm.org
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Subject: RE: [tcpm] rfc2581bis [2]: limiting cwnd inflation 
Date: Tue, 13 Feb 2007 23:50:34 -0800
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Thread-Topic: [tcpm] rfc2581bis [2]: limiting cwnd inflation 
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From: "Anantha Ramaiah \(ananth\)" <ananth@cisco.com>
To: <mallman@icir.org>
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=20
> -----Original Message-----
> From: mallman@icir.org [mailto:mallman@icir.org]=20
> Sent: Tuesday, February 13, 2007 9:36 PM
> To: Anantha Ramaiah (ananth)
> Cc: tcpm@ietf.org
> Subject: Re: [tcpm] rfc2581bis [2]: limiting cwnd inflation=20
>=20
> > Cwnd  =3D (f/2 + 3*SMSS) + 6SMSS > f.  This can happen since=20
> there seems=20
> > to be no explicit rule to cap the cwnd growth which may cause the=20
> > higher than appropriate sending rate to be used.  To make the above=20
> > example assuming f =3D 10 before fast recovery entered, we=20
> would be in a=20
> > situation where cwnd becomes 14!!, which is un-desirable.
>=20
> It is not.  After out iteration earlier today I went through=20
> this in more detail in my head.  You want cwnd to grow past=20
> where it was before recovery.  That is how you're able to=20
> strobe new segments out.  Simple
> example:
>=20
> FlightSize=3D40
> lose 1 segment; expect 39 dupacks (say none are lost) after=20
> the 3rd dupack:
>   cwnd =3D (FlightSize / 2) + 3 =3D 23
>   ssthresh =3D FlightSize / 2 =3D 20
>   retransmit
> receiver 36 more dupacks, on each:
>   cwnd +=3D 1
>   if FlightSize < cwnd:
>     transmit
>     FlightSize +=3D 1
> receive an ACK for everything
>   cwnd =3D ssthresh
>=20
> So, basically before the rexmt is ACKed cwnd will be 59 segments---or,
> 19 more than it was before the loss happened.  But, this=20
> increase is termed *artificial*.  We did not put 59 segments=20
> into the network during the last RTT, we put 19 segments into=20
> the network (or, half of the FlightSize when the loss=20
> occured)---so, we did in fact halve our rate even though the=20
> cwnd is larger.  Before continuing the cwnd collapses back to=20
> what it ought to be (20 segments in this case).

Ok, you are right on this, Mark. Basically the cwnd can grow up to a
maximum of (3f-2)/2 during fast recovery [ f denotes the flightsize
before entering FR]. Due to some interrmitent re-ordering in the
network, we could construct scenarios like below :-

Assume 40 segments sent out (f =3D 40), 1st segment got dropped. [same
example as above]
We get, lets say, 11 acks which fills the hole, which makes cwnd =3D 31
(20+11), f =3D 29(40 -11), after we still have remaining 28 acks, which
can slide the window one segment at a time [I am assuming that the
fast-retransmitted segment "overtook" the 28 segments . Of course I am
assuming some genuine networking scenarios like load balancing
scenarios/network conditions which causes some irregular ACK pattern
which can cause more than needed segments to be sent out. [ "needed
segments" =3D f/2 =3D 20 when we entered fast recovery]

Anyways, it appears that I may be trying hard to convince you on this,
but bottomline is we can construct scenarios.

>=20
> What the new text is attempting to do is to allow stacks to=20
> protect against malicious receivers that strobe many more=20
> dupacks into the stream to boost the sending rate. =20

Sure there is no issue with the existing text, but I was trying to make
it more generic.
Also it is not just mis-behaving receivers, another reason which I can
think of is some intermittent packet duplication can cause this as well.

> Regardless of the value of cwnd, I don't think there is a=20
> case whereby you'll be actually putting an inappropriate=20
> number of segments into the network when things are working properly.

Hmm.. This is what I am not too sure about, my loose example above tries
to demonstrate this. But in many real life scenarios partial ack's are
common which kicks in New Reno which has protection against window
deflation anyways since the cwnd gets reduced further.. May be it isn't
as worst as I think, I don't know.

-Anantha

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At 03:24 p.m. 13/02/2007, you wrote:

Comments inline....


>A new I-D has been posted on the IETF web site.
>
><http://www.ietf.org/internet-drafts/draft-mahesh-persist-timeout-00.txt>http://www.ietf.org/internet-drafts/draft-mahesh-persist-timeout-00.txt
>"TCP Maintenance and Minor Extensions", Mahesh Jethanandani, Murali 
>Bashyam, 9-Feb-07, <draft-mahesh-persist-timeout-00.txt> Comments are welcome.

What if I advertise a window of 1, instead?

Or, what if I advertise a window of zero, then before you abort the 
connection I advertise a window of a few bytes, and then I go back to 
advertising a window of zero (and so on)?

I think it is interesting to find a workaround for this type of 
resource exhaustion attack (as well as for Netkill, etc.).

However, I think the heuristics will need to be more complex. If not, 
it will be easy (and cheap) for the attacker to fool the proposed 
counter-measures. (the examples above are some possible ways to do so).

Kindest regards,

-- 
Fernando Gont
e-mail: fernando@gont.com.ar || fgont@acm.org
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From tcpm-bounces@ietf.org Wed Feb 14 03:14:36 2007
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From: MURALI BASHYAM <murali_bashyam@yahoo.com>
Subject: Re: [tcpm] New I-D (draft-mahesh-persist-timeout-00.txt)
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Good point.

This particular behaviour can be detected quite easily
when compared to a well-behaved receiver. A
well-behaved TCP receiver will do receive side silly
window avoidance, and would only advertise a window
increase when at least a MSS worth of buffer space is
available (typically). 

Murali

--- Fernando Gont <fernando@gont.com.ar> wrote:

> At 03:24 p.m. 13/02/2007, you wrote:
> 
> Comments inline....
> 
> 
> >A new I-D has been posted on the IETF web site.
> >
>
><http://www.ietf.org/internet-drafts/draft-mahesh-persist-timeout-00.txt>http://www.ietf.org/internet-drafts/draft-mahesh-persist-timeout-00.txt
> >"TCP Maintenance and Minor Extensions", Mahesh
> Jethanandani, Murali 
> >Bashyam, 9-Feb-07,
> <draft-mahesh-persist-timeout-00.txt> Comments are
> welcome.
> 
> What if I advertise a window of 1, instead?
> 
> Or, what if I advertise a window of zero, then
> before you abort the 
> connection I advertise a window of a few bytes, and
> then I go back to 
> advertising a window of zero (and so on)?
> 
> I think it is interesting to find a workaround for
> this type of 
> resource exhaustion attack (as well as for Netkill,
> etc.).
> 
> However, I think the heuristics will need to be more
> complex. If not, 
> it will be easy (and cheap) for the attacker to fool
> the proposed 
> counter-measures. (the examples above are some
> possible ways to do so).
> 
> 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
> 
> 
> 
> 
> 
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www1.ietf.org/mailman/listinfo/tcpm
> 



 
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Subject: RE: [tcpm] New I-D (draft-mahesh-persist-timeout-00.txt)
Date: Wed, 14 Feb 2007 00:35:59 -0800
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From: "Anantha Ramaiah \(ananth\)" <ananth@cisco.com>
To: "MURALI BASHYAM" <murali_bashyam@yahoo.com>,
	"Fernando Gont" <fernando@gont.com.ar>,
	"Mahesh Jethanandani \(mahesh\)" <mahesh@cisco.com>
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Yep, on a related note, some TCP stacks shrink windows, window shrinking
is used as a convenience tool in some cases :-) We probably need to deal
with such cases as well, since the robustness principle allows it.

-Anantha=20

> -----Original Message-----
> From: MURALI BASHYAM [mailto:murali_bashyam@yahoo.com]=20
> Sent: Wednesday, February 14, 2007 12:14 AM
> To: Fernando Gont; Mahesh Jethanandani (mahesh)
> Cc: tcpm@ietf.org
> Subject: Re: [tcpm] New I-D (draft-mahesh-persist-timeout-00.txt)
>=20
> Good point.
>=20
> This particular behaviour can be detected quite easily when=20
> compared to a well-behaved receiver. A well-behaved TCP=20
> receiver will do receive side silly window avoidance, and=20
> would only advertise a window increase when at least a MSS=20
> worth of buffer space is available (typically).=20
>=20
> Murali
>=20
> --- Fernando Gont <fernando@gont.com.ar> wrote:
>=20
> > At 03:24 p.m. 13/02/2007, you wrote:
> >=20
> > Comments inline....
> >=20
> >=20
> > >A new I-D has been posted on the IETF web site.
> > >
> >
> ><http://www.ietf.org/internet-drafts/draft-mahesh-persist-tim
> eout-00.tx
> >t>http://www.ietf.org/internet-drafts/draft-mahesh-persist-ti
> meout-00.t
> >xt
> > >"TCP Maintenance and Minor Extensions", Mahesh
> > Jethanandani, Murali
> > >Bashyam, 9-Feb-07,
> > <draft-mahesh-persist-timeout-00.txt> Comments are welcome.
> >=20
> > What if I advertise a window of 1, instead?
> >=20
> > Or, what if I advertise a window of zero, then before you abort the=20
> > connection I advertise a window of a few bytes, and then I=20
> go back to=20
> > advertising a window of zero (and so on)?
> >=20
> > I think it is interesting to find a workaround for this type of=20
> > resource exhaustion attack (as well as for Netkill, etc.).
> >=20
> > However, I think the heuristics will need to be more=20
> complex. If not,=20
> > it will be easy (and cheap) for the attacker to fool the proposed=20
> > counter-measures. (the examples above are some possible ways to do=20
> > so).
> >=20
> > Kindest regards,
> >=20
> > --
> > Fernando Gont
> > e-mail: fernando@gont.com.ar || fgont@acm.org PGP Fingerprint: 7809=20
> > 84F5 322E 45C7 F1C9 3945 96EE A9EF D076 FFF1
> >=20
> >=20
> >=20
> >=20
> >=20
> > _______________________________________________
> > tcpm mailing list
> > tcpm@ietf.org
> > https://www1.ietf.org/mailman/listinfo/tcpm
> >=20
>=20
>=20
>=20
> =20
> ______________________________________________________________
> ______________________
> 8:00? 8:25? 8:40? Find a flick in no time with the Yahoo!=20
> Search movie showtime shortcut.
> http://tools.search.yahoo.com/shortcuts/#news
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www1.ietf.org/mailman/listinfo/tcpm
>=20

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From tcpm-bounces@ietf.org Wed Feb 14 03:59:19 2007
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From: Fernando Gont <fernando@gont.com.ar>
Subject: RE: [tcpm] New I-D (draft-mahesh-persist-timeout-00.txt)
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At 05:35 a.m. 14/02/2007, Anantha Ramaiah \(ananth\) wrote:

>Yep, on a related note, some TCP stacks shrink windows, window shrinking
>is used as a convenience tool in some cases :-) We probably need to deal
>with such cases as well, since the robustness principle allows it.

The amount of data queued for transmission, is another factor worth 
considering in the heuristics, as for DoS attacks there will usually 
be a large amount of queued data (send buffer full, or close to).

Also, if the corresponding socket has been close()ed by the 
application running at the TCP sender, the application will usually 
have no control on the socket anymore (unless the SO_LINGER (?) 
socket option was set?). This may be worth noting in the doc, too.

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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From tcpm-bounces@ietf.org Wed Feb 14 09:05:50 2007
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To: "Anantha Ramaiah (ananth)" <ananth@cisco.com>
From: Mark Allman <mallman@icir.org>
Subject: Re: [tcpm] rfc2581bis [2]: limiting cwnd inflation 
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> Assume 40 segments sent out (f = 40), 1st segment got dropped. [same
> example as above]
> We get, lets say, 11 acks which fills the hole, which makes cwnd = 31
> (20+11), f = 29(40 -11), after we still have remaining 28 acks, which
> can slide the window one segment at a time [I am assuming that the
> fast-retransmitted segment "overtook" the 28 segments . Of course I am
> assuming some genuine networking scenarios like load balancing
> scenarios/network conditions which causes some irregular ACK pattern
> which can cause more than needed segments to be sent out. [ "needed
> segments" = f/2 = 20 when we entered fast recovery]

I don't understand this.  As soon as the window slides we are out of
fast recovery, the cwnd collapses back down and we go to congestion
avoidance to govern sending instead of the fast recovery rules.  How
does the above lead us to sending inappropriately.

> Also it is not just mis-behaving receivers, another reason which I can
> think of is some intermittent packet duplication can cause this as
> well.

Yep.  And, I willing to add a parenthetical or something about this
point.  But, for this to be a problem you'd have to have a *lot* of
duplication (which runs against all the evidence about that phenomenon
that I have seen).  I.e., I am not worried about a couple extra dupacks
throwing us off a little.  The concern here is that mis-behavior can
cause many times more segments than appropriate to be sent.

> > Regardless of the value of cwnd, I don't think there is a case
> > whereby you'll be actually putting an inappropriate number of
> > segments into the network when things are working properly.
> 
> Hmm.. This is what I am not too sure about, 

Show us how.

(I am not saying there is not a scenario.  I just haven't yet seen a
scenario that would cause a big problem in the absence of a mis-behaving
receiver.  (Or, a very very flawed network element that is duplicating
packets like nobody's business.))

allman




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To: Fernando Gont <fernando@gont.com.ar>
From: Mark Allman <mallman@icir.org>
Subject: Re: [tcpm] Some comments on 2581bis 
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> ** RFC 2581 & SACK
> Page 3 says:
>          Alternatively, a TCP that utilizes selective acknowledgments
>          [RFC2018,RFC2883] can determine an incoming ACK is a "duplicate"
>          if the ACK contains previously unknown SACK information.
> 
> And page 7 says:
> Note that a sender using SACK [RFC2018]
>          MUST NOT send new data unless the incoming duplicate
>          acknowledgment contains new SACK information.
> 
> IMHO, if you keep the "MUST NOT" in page 7, you'd probably
> s/can/MUST/ in page 3. Also, maybe it would be worth to clarify that
> the SACK information should be "in window"?

Well, duplicate ACKs in 2581 are not always to send out new data.

That said, I am wondering if we should just nuke the sentence you quote
above.  This document does not deal with SACK-based loss recovery and if
you have SACK then you should be doing something smarter than this
document.  And, so maybe we don't need to really define things about
SACK-based loss recovery in here.  Certainly, I don't think we want to
get into whether SACK blocks that this document doesn't use at all are
in window or not.

> ** Limits on dupacks
> Would it make sense to limit the number of accepted dupacks to
> Max_dupacks = (Flight_size_fr - 1) / SMSS ?
> where Flight_size_fr is the Flight_Size at the point fast retransmit
> is triggered.

I'd like to hear what people think on this notion.  The intent in
putting the new text in was not to tightly specify a mechanism, but to
note that there is good reason to put some sort of limit on things.  How
that is done does not seem important to me.  (*If* it is done only seems
marginally helpful to me.)  I.e., one could envision a cwnd limit or
trying (because we don't keep track in some stacks) to track the number
of packets vs. dupacks as above or ...

I guess I would rather leave it as sort-of fuzzy guidance and not
mechanism or just drop it as new functionality that is out of scope for
the revision.

What do folks think?

allman




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Mark Allman wrote:
>> ** RFC 2581 & SACK
>> Page 3 says:
>>          Alternatively, a TCP that utilizes selective acknowledgments
>>          [RFC2018,RFC2883] can determine an incoming ACK is a "duplicate"
>>          if the ACK contains previously unknown SACK information.
>>
>> And page 7 says:
>> Note that a sender using SACK [RFC2018]
>>          MUST NOT send new data unless the incoming duplicate
>>          acknowledgment contains new SACK information.
>>
>> IMHO, if you keep the "MUST NOT" in page 7, you'd probably
>> s/can/MUST/ in page 3. Also, maybe it would be worth to clarify that
>> the SACK information should be "in window"?
> 
> Well, duplicate ACKs in 2581 are not always to send out new data.
> 
> That said, I am wondering if we should just nuke the sentence you quote
> above.  This document does not deal with SACK-based loss recovery and if
> you have SACK then you should be doing something smarter than this
> document.  And, so maybe we don't need to really define things about
> SACK-based loss recovery in here.  Certainly, I don't think we want to
> get into whether SACK blocks that this document doesn't use at all are
> in window or not.

The notion of the sender "using" SACK is a bit fuzzy anyway, since some 
stacks will send the options but not process them in any meaningful way 
(writing scoreboard code is hard).

   -John

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At 12:05 p.m. 14/02/2007, Mark Allman wrote:

> > ** RFC 2581 & SACK
> > Page 3 says:
> >          Alternatively, a TCP that utilizes selective acknowledgments
> >          [RFC2018,RFC2883] can determine an incoming ACK is a "duplicate"
> >          if the ACK contains previously unknown SACK information.
> >
> > And page 7 says:
> > Note that a sender using SACK [RFC2018]
> >          MUST NOT send new data unless the incoming duplicate
> >          acknowledgment contains new SACK information.
> >
> > IMHO, if you keep the "MUST NOT" in page 7, you'd probably
> > s/can/MUST/ in page 3. Also, maybe it would be worth to clarify that
> > the SACK information should be "in window"?
>
>Well, duplicate ACKs in 2581 are not always to send out new data.

My point is: if you disregard "dupacks" that don't have new sack 
information for sending new data, why would you let them trigger loss recovery?



>That said, I am wondering if we should just nuke the sentence you quote
>above.  This document does not deal with SACK-based loss recovery and if
>you have SACK then you should be doing something smarter than this
>document.  And, so maybe we don't need to really define things about
>SACK-based loss recovery in here.  Certainly, I don't think we want to
>get into whether SACK blocks that this document doesn't use at all are
>in window or not.

I fully agree with this view.



> > ** Limits on dupacks
> > Would it make sense to limit the number of accepted dupacks to
> > Max_dupacks = (Flight_size_fr - 1) / SMSS ?
> > where Flight_size_fr is the Flight_Size at the point fast retransmit
> > is triggered.
>
>I'd like to hear what people think on this notion.  The intent in
>putting the new text in was not to tightly specify a mechanism, but to
>note that there is good reason to put some sort of limit on things.  How
>that is done does not seem important to me.  (*If* it is done only seems
>marginally helpful to me.)  I.e., one could envision a cwnd limit or
>trying (because we don't keep track in some stacks) to track the number
>of packets vs. dupacks as above or ...
>
>I guess I would rather leave it as sort-of fuzzy guidance and not
>mechanism or just drop it as new functionality that is out of scope for
>the revision.

I think it would be helpful to describe possible mechanisms (such as 
the one above). If not, chances are that people is not going to 
implement it, or they may get the limit wrong. The doc already some 
includes text about ack-division attacks, so IMHO I don't think 
describing a mechanism for enforcing limits on dupacks would be out of scope.

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 Wed Feb 14 12:03:35 2007
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> I think it would be helpful to describe possible mechanisms (such as
> the one above). If not, chances are that people is not going to
> implement it, or they may get the limit wrong. The doc already some
> includes text about ack-division attacks, so IMHO I don't think
> describing a mechanism for enforcing limits on dupacks would be out of
> scope.

The hesitation here is that we're making things up for inclusion in a
document that we are trying to progress to draft standard.  I am a
little uncomfortable with that.  The other things in the document have
some more long-term understanding and study behind them.

I don't mind the document saying "here is a case where you might want to
be more conservative than allowed".  But, if we translate that into one
or more mechanisms we could easily get bit.  I have done this "it is
such an easy tweak, how could it be wrong?" business enough times over
the years to know that the interactions are tricky and subtle and it
takes work to understand the implications of these things.  And,
sometimes the "easy, for sure gotta be right" changes are absolutely
wrong.  I am just feeling like we don't want to do that in this
document. 

Just my opinion ...

allman




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To: John Heffner <jheffner@psc.edu>
From: Mark Allman <mallman@icir.org>
Subject: Re: [tcpm] Some comments on 2581bis 
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> The notion of the sender "using" SACK is a bit fuzzy anyway, since
> some stacks will send the options but not process them in any
> meaningful way (writing scoreboard code is hard).

This is getting a little better.  Contrast the 2000 and 2004 results on
actual SACK usage (not just sending them) in the paper below.  Still not
stunning in 2004, but at least it was on the rise and hopefully it has
continued to be.

allman



Alberto Medina, Mark Allman, Sally Floyd.  Measuring the Evolution of
Transport Protocols in the Internet.  ACM Computer Communication Review,
35(2), April 2005. 
http://www.icir.org/mallman/papers/tcp-evo-ccr05.ps





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I am sending the comments which we received from Alfred to the wider =
audience. I have gone through that and most of them look acceptable. =
Revision 07 which will posted very soon would incoporate these comments =
among other things. I'll also broadcast the response which is being sent =
to Alfred.

Thanks,
-Anantha

-----Original Message-----
From: Alfred H=CEnes [mailto:ah@tr-sys.de]=20
Sent: Wednesday, January 17, 2007 10:22 AM
To: Anantha Ramaiah (ananth); Randall Stewart (rrs); Mitesh Dalal =
(mdalal)
Subject: draft-ietf-tcpm-tcpsecure-06

Hello,
after studying the Internet-Draft authored/edited by you,
    draft-ietf-tcpm-tcpsecure-06,
I'd like to submit a few comments, addressing the many textual/editorial =
flaws I found in that memo.

The items below are presented in textual order.
To give more context, sometimes I quote larger blocks of text literally =
and show the proposed replacement in 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 porposals.


(0)  obsolete Ref.

[RFC1771] has been obsoleted, one year ago, by RFC 4271, January 2006.

Please update the multiple references.  Below, I only include the =
appropriate changes for snippits presented for other reasons as well.


(1)  Abstract, end of 1st paragraph

                                            [...].  A combination of
|  increasing window sizes and applications using a longer term =20
| connections (e.g.  H-323 or Border Gateway Protocol [RFC1771]) have
   left modern TCP implementation more vulnerable to these types of
   spoofed packet injection attacks.
---
                                            [...].  A combination of
|  increasing window sizes and applications using longer term =20
| connections (e.g., H-323 or Border Gateway Protocol [RFC1771]) have
   left modern TCP implementation more vulnerable to these types of
   spoofed packet injection attacks.


(2)  Section 1, 1st paragraph

   TCP [RFC0793] is widely deployed and the most common reliable end to
   end transport protocol used for data communication in todays
   Internet.  Yet when it was defined over 20 years ago the Internet, as
   we know it, was a different place lacking many of the threats that
   are now common.  TCP spoofing attacks are one such attack that are
   seen on the Internet today.
---
   TCP [RFC0793] is widely deployed and the most common reliable end to
|  end transport protocol used for data communication in today's =20
| Internet.  Yet when it was defined over 30 years ago, the Internet, =20
| as we know it, was a different place, lacking many of the threats
   that are now common.  TCP spoofing attacks are one such attack that
   are seen on the Internet today.

[Note: The first TCP Specification published as an RFC was RFC 675,  =
December 1974, obsoleted by RFC 761 (January 1980), and finally  RFC 793 =
=3D STD 7 (September 1981).  According to Jon Postel, there  have been =
at 7 earlier editions of the TCP specification not  circulated as RFCs.
 So don't tell a ripe, 32+ years ol' TCP a teenager ! ]


(3)  Section 1.1=20

In bullet 1):

      ... number of factors most notably ...
---
      ... number of factors, most notably ...
                           ^
In the last paragraph:

      ... much more difficult then ...
---
      ... much more difficult than ...
                                ^

(4)  Section 1.2

Typos in the 5th paragraph:

   For the purposes of the rest of this discussion we will assume that
   the attacker knows the 4-tuple values.  This assumption will help us
|  focus on the effects of the window size verses the number of TCP
   packets an attacker must generate.  This assumption will rarely be
   true in the real Internet since at least the client port number will
   provide us with some amount of randomness (depending on operating
   system).
---
   For the purposes of the rest of this discussion we will assume that
   the attacker knows the 4-tuple values.  This assumption will help us
|  focus on the effects of the window size versus the number of TCP
   packets an attacker must generate.  This assumption will rarely be
   true in the real Internet since at least the client port number will
|  provide us with some amount of randomness (depending on the operating
   system).

In the 6th paragraph, I recommend to replace the uppercase 'X' by a =
lowercase 'x' as a multiplication sign, or use '*' instead.
Simplification by evaluation of the [constant parts of the] terms =
("constant folding") might be considered as well.

At the end of the 8th paragraph:

       [...].  Resetting the connection can result in medium term
   unavailability due to the need to rebuild routing tables and route
   flapping see [NISCC] for further details.
---
       [...].  Resetting the connection can result in medium term
   unavailability due to the need to rebuild routing tables and route
|  flapping; see [NISCC] for further details.
           ^
9th paragraph:

   It also needs to be emphasized that for applications like BGP which
   use the TCP MD5 option [RFC2385], can make the attacks described in
   this specification much harder to accomplish.
---
|  It also needs to be emphasized that, for applications like BGP, use =20
| of the TCP MD5 option [RFC2385] can make the attacks described in
   this specification much harder to accomplish.
or:
|  It also needs to be emphasized that for applications like BGP using =20
| the TCP MD5 option [RFC2385], the attacks described in this =20
| specification are much harder to accomplish.

10th (=3D last) paragraph:
                           vvvvvvvvvv                              vv?
|  It should be noted that there are existing alternative protection
   against the threats that this document addresses.  [...]
---                                                                 v
|  It should be noted that there are existing alternative protections
   against the threats that this document addresses.  [...]


(5)  Section 3.2 -- multiple typos

Below bullet '2)' :

   Instead, this document requires that implementations MUST perform the
   following steps in place of those specified in [RFC0793](listed
   above).
---                                                       ^^
   Instead, this document requires that implementations MUST perform the
|  following steps in place of those specified in [RFC0793] (as listed
   above).
                                                           ^^^^^ [ For =
additional clarification, I have added the word "as". ]

In bullet 'C)'

      The previous text,quoted from [RFC0793] ...
---
      The previous text, quoted from [RFC0793] ...
                        ^
and subsequently:
                                    vvvvvvv
|                        [...].  If the the RST arrives and its sequence
      number field does NOT match the next expected sequence number but
      is within the window, then the receiver should generate an ACK.
---
|                        [...].  If the RST arrives and its sequence
      number field does NOT match the next expected sequence number but
      is within the window, then the receiver should generate an ACK.

and finally, at the end of the bullet:

                                          [...].  This second RST if it
      reaches the sender will cause the connection to be aborted since
      the sequence number would now be an exact match.
---
|                                         [...].  This second RST, if it
|     reaches the sender, will cause the connection to be aborted since
      the sequence number would now be an exact match.
or:
|                                         [...].  If this second RST
|     reaches the sender, it will cause the connection to be aborted
      since the sequence number would now be an exact match.

And at the end of the section, add the missing full stop :

      [...].  This concern is discussed in Section 9.
                                                    ^

(6)  Section 4.1
  vv                                             v
   Analysis of the reset attack using the RST bit, highlights another
   possible avenue for a blind attacker using a similar set of sequence
   number guessing.  [...]
---
|  The Analysis of the reset attack using the RST bit highlights another
   possible avenue for a blind attacker using a similar set of sequence
   number guessing.  [...]


(7)  Section 4.2

In the 2nd paragraph after bullet 'A)' :

                     [...].  Upon receipt of a valid RST, the local TCP
|  endpoint MUST terminate its connection.  The local TCP endpoint
   should then rely on SYN retransmission from the remote end to re-
   establish the connection.
---
                     [...].  Upon receipt of a valid RST, the local TCP
|  endpoint MUST terminates its connection.  The local TCP endpoint
   should then rely on SYN retransmission from the remote end to re-
   establish the connection.

In the 4th paragraph after bullet 'A)' , change  "non spoofed SYN"  -->  =
"non-spoofed SYN" .

And at the end of the section, add the missing full stop :

      [...].  This concern is discussed in Section 9.
                                                    ^

(8)  Section 5.1

Typos and mismatched parentheses in the 1st paragraph:

   A third type of attack is also highlighted by both the RST and SYN
   attacks.  It is also possible to inject data into a TCP connection by
|  simply guessing the a sequence number within the current receive
   window of the victim.  The ACK value of any data segment is
   considered valid as long as it does not acknowledge data ahead of the
   next segment to send.  In other words an ACK value is acceptable if
|  it is (SND.UNA-(2^31-1)) <=3D SEG.ACK <=3D SND.NXT).  This means that =
an
   attacker has to guess two ACK values with every guessed sequence
|  number so that the chances successfully injecting data into a
   connection are 1 in ((2^32 / RCV.WND) * 2).
---
   A third type of attack is also highlighted by both the RST and SYN
   attacks.  It is also possible to inject data into a TCP connection by
|  simply guessing a sequence number within the current receive window
   of the victim.  The ACK value of any data segment is considered valid
   as long as it does not acknowledge data ahead of the next segment to
   send.  In other words an ACK value is acceptable if it is
|  ((SND.UNA-(2^31-1)) <=3D SEG.ACK <=3D SND.NXT).  This means that an
   attacker has to guess two ACK values with every guessed sequence
|  number so that the chances of successfully injecting data into a
   connection are 1 in ((2^32 / RCV.WND) * 2).


(9)  Section 5.2

Typos, and improved wording, in the 1st paragraph:

   An additional input check MUST be added to any incoming segment.  The
   ACK value MUST be acceptable only if it is in the range of ((SND.UNA
|  - MAX.SND.WND) <=3D SEG.ACK <=3D SND.NXT).A new state variable
   MAX.SND.WND is defined as the largest window that the local sender
|  has ever received from its peer.  This window may be a scaled value =20
| i.e. the value may be larger than 65,535 bytes ([RFC1323]).  This
   small check will reduce the vulnerability to an attacker guessing a
   valid sequence number since not only must he/she guess the sequence
   number in window, but must also guess a proper ACK value within a
   scoped range.  This mitigation reduces but does not eliminate the
   ability to generate false segments.  It does however reduce the
   probability that invalid data will be injected.
---
   An additional input check MUST be added to any incoming segment.  The
   ACK value MUST be acceptable only if it is in the range of ((SND.UNA
|  - MAX.SND.WND) <=3D SEG.ACK <=3D SND.NXT).  A new state variable
   MAX.SND.WND is defined as the largest window that the local sender
|  has ever received from its peer.  This window may be scaled to a =20
| value larger than 65,535 bytes ([RFC1323]).  This small check will
   reduce the vulnerability to an attacker guessing a valid sequence
|  number, since he/she not only must guess the in-window sequence =20
| number, but also a proper ACK value within a scoped range.  This =20
| mitigation reduces, but does not eliminate, the ability to generate
   false segments.  It does, however, reduce the probability that
   invalid data will be injected.

And again, at the end of the section, add the missing full stop :

      [...].  This concern is discussed in Section 9.
                                                    ^

(10)  Section 6

2nd paragraph after the numbered bullets:
                                    vvv
|  An implementation SHOULD include a ACK throttling mechanism to be
   conservative.  [...]
---                                 vvvv
|  An implementation SHOULD include an ACK throttling mechanism to be
   conservative.  [...]


(11)  Section 8.1

1st paragraph:

   Consider a middlebox M-B tracking connection between two TCP endhosts
   E-A and E-C.  If E-C sends a RST with a sequence number that is
   within the window but not an exact match to reset the connection and
   M-B does not have the fix recommended in this document, it may clear
   the connection and forward the RST to E-A saving an incorrect
   sequence number.  [...]
---                                            v
|  Consider a middlebox M-B tracking connections between two TCP
   endhosts E-A and E-C.  If E-C sends a RST with a sequence number that
   is within the window but not an exact match to reset the connection
   and M-B does not have the fix recommended in this document, it may
   clear the connection and forward the RST to E-A saving an incorrect
   sequence number.  [...]

and:
                             vv          vv   v
             [...].  This RST, will again, not acceptable and may
   trigger a challenge ACK.
---                          v          v    vv
             [...].  This RST will again not be acceptable and may
   trigger a challenge ACK.

2nd paragraph:

   The above situation may result in a RST/ACK war.  However we believe
   that if such a case exists in the Internet the middle box design does
   not comply to [RFC0793].  [...]
---
|  The above situation may result in a RST/ACK war.  However, we believe =
=20
| that if such a case exists in the Internet, the middle box design
   does not comply to [RFC0793].  [...]


(12)  Section 8.2

1st paragraph:

                      [...].  The scenario is the same as the earlier
   case, but in this case instead of sending the cached RST, the
   middlebox (M-B) sends a RST that computes its sequence number as a
   sum of the ack field in the ACK and the window advertised by the ACK
   that was sent by E-A to challenge the RST as depicted below.  [...]
---
                      [...].  The scenario is the same as the earlier
   case, but in this case instead of sending the cached RST, the
|  middlebox (M-B) sends a RST that computes its sequence number as the
   sum of the ack field in the ACK and the window advertised by the ACK
   that was sent by E-A to challenge the RST as depicted below.  [...]


(13)  Section 9, 5th paragraph

   Another consideration that should be noted is a reflector attack is
   possible with the required RST/SYN mitigation techniques.  In this
   attack an off-path attacker can cause a victim to send an ACK segment
   for each spoofed RST/SYN segment that lies within the current receive
   window of the victim.  [...]
---
|  Another consideration that should be noted is a reflector attack, =20
| which is possible with the required RST/SYN mitigation techniques.
|  In this attack, an off-path attacker can cause a victim to send an
   ACK segment for each spoofed RST/SYN segment that lies within the
   current receive window of the victim.  [...]


Best regards,
  Alfred H=CEnes.

--=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                     |
+------------------------+--------------------------------------------+

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A New Internet-Draft is available from the on-line Internet-Drafts 
directories.
This draft is a work item of the TCP Maintenance and Minor Extensions Working Group of the IETF.

	Title		: TCP Congestion Control
	Author(s)	: M. Allman, et al.
	Filename	: draft-ietf-tcpm-rfc2581bis-02.txt
	Pages		: 16
	Date		: 2007-2-15
	
This document defines TCP's four intertwined congestion control
    algorithms: slow start, congestion avoidance, fast retransmit, and
    fast recovery.  In addition, the document specifies how TCP should
    begin transmission after a relatively long idle period, as well as
    discussing various acknowledgment generation methods.

A URL for this Internet-Draft is:
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Hi folks!

I am looking for a little bit of implementation experience on 2581.  I
should really keep better track of this sort of thing, but if anyone out
there is a TCP implementer and could contact me off-list, I'd appreciate
it.

Thanks,
allman




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All,=20

As you will shortly see we have just posted a new ID called "A  TCP Test
To Allow Senders to Identify Receiver Cheating"
draft-moncaster-tcpm-rcv-cheat-00. This draft was written as a result of
a request made to Bob Briscoe at IETF 67. There he mentioned to TVWG
that we would be happy to produce a transport layer nonce to protect
against receivers concealing lost packets. This suggestion was broadly
welcomed by the WG and this draft is our response. Initially we intended
to submit this to TSVWG as they requested it. However after checking
with the co-chairs of both TCPM and TSVWG it was agreed that TCPM would
be the best forum for the draft. The draft is available in various
formats at
http://www.cs.ucl.ac.uk/staff/B.Briscoe/pubs.html#tcp-rcv-cheat.

After some background research we decided that an actual nonce would
probably be impractical since it would require modifications to both
sender and receiver. This would make it very hard to enforce as the
receiver could simply pretend to be a legacy one that didn't support the
new nonce. We were keen that our system should be robust and should
require the receiver simply to behave as mandated by the existing TCP
protocols. Indeed the idea was that the solution should actually
leverage the existing TCP behaviours in order to test the honesty of the
receiver. We decided to base our new solution on a solution by Rob
Sherwood et al presented in "Misbehaving TCP receivers can cause
Internet-wide congestion collapse". Their solution, called the randomly
skipped segments solution, suggests purposely removing a segment from
the transmitted data stream and only transmitting it after the receiver
sends a NACK for that segment. If in the meantime the receiver sends an
ACK for a later segment then they are deemed to be behaving dishonestly.

A recent thread on the TCPM mailing list (starting 11th January 2007
"DoS attack from misbehaving receivers") has been discussing the
randomly skipped segments solution in some detail and reveals that there
is a lot of unease about such a test. Our solution is hopefully more
palatable and consists of 2 stages. In the first stage we
probabilistically test the honesty of a receiver. To do this we select a
segment N and a displacement D. Instead of transmitting segments N-1, N,
N+1, N+2, etc we transmit N-1, N+1, N+2, ..., N+D, N, ... An honest TCP
receiver should respond to this missing data by generating a stream of
duplicate ACKs for segment N-1. Any receiver that doesn't do so is
probably behaving dishonestly but, because TCP doesn't guarantee
reliable delivery of ACKs we can't state this for certain. We therefore
state that any receiver that fails this test should be subjected to a
further test that is essentially the same as the Sherwood test. It is
felt that the first test causes minimal damage to an honest receiver and
is essentially compliant with the TCP protocol. The second test is then
reserved for those receivers that we are pretty certain are behaving
dishonestly. We have currently left it up to the community to decide how
to sanction any receiver that fails the second test. We suggest the
appropriate response might be to terminate the connection but others
might instead feel that other sanctions such as reducing the congestion
window and slow start threshold to 1 would be more appropriate.

Both our tests force accurate reporting of losses which was our initial
aim. They also both make it much harder for a receiver to optimistically
ACK data which is an added benefit given the possible harm such
optimistic ACKs might cause.

We will be presenting this at IETF 68 in Prague and would welcome any
comments from people as to the utility of our solution and any
improvements that could be made.

Toby Moncaster=20
Future Communications Architecture
Networks Research Centre
BT Group Chief Technology Office
_________________________________________________
t: (+44) (0) 1473 648734
m: (+44) (0) 7764 185416=20
________________________________________________________________________
___

Notice: This contribution is the personal view of the author and does
not necessarily reflect the technical nor commercial direction of BT
plc.


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To: weddy@grc.nasa.gov
From: Mark Allman <mallman@icir.org>
Subject: Re: [tcpm] New I-D 
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Regarding:
  http://www.ietf.org/internet-drafts/draft-mahesh-persist-timeout-00.txt

Wes:
> [...] but it looks to me like this is more of an OS mechanism to
> manage resources rather than a protocol-based mechanism.  Is this
> correct? 

I'm with Wes here.

I just took a spin through the draft and am left with three thoughts ...

  + You say apps don't have a good role to play even after citing apps
    that 'pause' as the culprit.  I can't make these things agree in my
    head.  Even in the absence of TCP-specific information it seems that
    these apps could time themselves out if no progress is being made
    (i.e. data getting exchanged).

    (Further, things like HTTP sends after which the app goes away and
    then getting wedged into persist don't worry me much.  That sounds
    like a once in a blue moon sort of situation to me.)

  + This seems fundamentally like an operating system resource issue to
    me and not something for a networking standards body.  OSes deal
    with all sorts of resource contention issues without standards.  Why
    does this problem need to be solved the same way by each OS?

  + In particular, I think you have solved the problem wrong.  Don't get
    me wrong ... maybe it is perfectly OK for your purposes and it is
    fine with me if you want to use it.  However, I would want something
    a little smarter to really avoid the mythical attacks you describe.
    All you did was to cap the length of connections.  So, conceivably a
    periodic attack could still keep all your resources busy.  It seems
    to me that you'd want to use some sort of scheme that actually took
    into account contention and resource exhaustion.  E.g., some sort of
    LRU what-has-been-in-persist longest when you need new resources and
    they are not available sort of culling scheme?  Or, when you have
    exhausted X% of your resources to keep some in reserve.  That might
    be wrong and busted itself (it is quickly off the top of my head),
    but it at least takes into account the resource contention unlike a
    naive timeout.

Just my two bits ...

allman




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From tcpm-bounces@ietf.org Tue Feb 20 14:37:39 2007
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Date: Tue, 20 Feb 2007 14:36:52 -0500
From: Rob Sherwood <capveg@cs.umd.edu>
To: toby.moncaster@bt.com
Subject: Re: [tcpm] New I-D posted :  draft-moncaster-tcpm-rcv-cheat-00
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On Tue, Feb 20, 2007 at 05:27:03PM -0000, toby.moncaster@bt.com wrote:
> All, 
> 
> As you will shortly see we have just posted a new ID called "A  TCP Test
> To Allow Senders to Identify Receiver Cheating"
> draft-moncaster-tcpm-rcv-cheat-00. This draft was written as a result of
> a request made to Bob Briscoe at IETF 67. There he mentioned to TVWG
> that we would be happy to produce a transport layer nonce to protect
> against receivers concealing lost packets. This suggestion was broadly
> welcomed by the WG and this draft is our response. Initially we intended
> to submit this to TSVWG as they requested it. However after checking
> with the co-chairs of both TCPM and TSVWG it was agreed that TCPM would
> be the best forum for the draft. The draft is available in various
> formats at
> http://www.cs.ucl.ac.uk/staff/B.Briscoe/pubs.html#tcp-rcv-cheat.

Hi Toby,

I gave a quick read through the draft on your website, and have a few
comments.


Section 5.1) 

I think there might be some confusion here, but the skipped segment
solution that we proposed in our paper does not require the receiver
supports SACK.  Trivially, our paper presents performance results for
receivers that do and do not support SACK.  If the sender skips segment
N, any ACK > N indicates misbehavior; SACK is simply not involved
for correctness.  If SACK is not supported in the connection, then the
standard performance penalties for not implementing SACK apply, i.e.,
the receiver has to duplicate ACK each loss in turn.  Am I missing
something here?

Section 6.2.4)

"Periodically and randomly, any heavily loaded TCP sender SHOULD check
the honesty of its receivers using the probabilistic test."

I don't think it is sufficient that only heavily loaded TCP senders
perform this test.  Attackers can simply draw a small amount of traffic
from many ("jillions" to quote Mark :-) of servers, not loading any one,
but still have a large aggregate amount of traffic.  In other words,
if the solution is only applied to loaded servers, then the attack
still exists.



At a high level, I like the proposed two phase test in that it is
slightly more efficient then the randomly skipped segment test alone,
i.e., instead of honest receivers being penalized 1 RTT every test,
they are only penalized the buffer time and 1 RTT if network reordering
(pathologically?) undoes the reordering test.  It is not clear, to
me at least, how the efficiency gain weighs against the additional
complexity of code, but presumably others can comment on that.  However,
my understanding of the concern raised on this list (if I can summarize
other's opinions) was that there was a philosophical objection to mucking
up[1] the TCP code for an attack which is not being actively exploited.
In other words, it wasn't the efficiency of the solution that was at
issue, but whether is should be applied at all.

Should people decide that this two phases approach is preferable, it
might be possible to simplify the first phase test as follows.  Rather
then reordering segments N-1,N,N+1,..,N+D to N-1,N+1,..,N+D,N as Toby
et. al suggest, it might be better to more frequently perform this test
with D=1, (i.e., simply swap the order of two sequential segments) and
after some threshold number K failures, apply the second phase randomly
skipped segment test.   I only suggest this as the analysis is probably
slightly simpler in that the reordering rate of pairs of packets have been
previously measured, where as the probability that the N-1,N,N+1,..,N+D
to N-1,N+1,..,N+D,N reordering would be undone is less well studied.


- Rob
.

[1] "mucking up" is clearly a technical term :-)

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From: MURALI BASHYAM <murali_bashyam@yahoo.com>
Subject: Re: [tcpm] New I-D 
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--- Mark Allman <mallman@icir.org> wrote:

> 
> Regarding:
>  
>
http://www.ietf.org/internet-drafts/draft-mahesh-persist-timeout-00.txt
> 
> Wes:
> > [...] but it looks to me like this is more of an
> OS mechanism to
> > manage resources rather than a protocol-based
> mechanism.  Is this
> > correct? 
> 
> I'm with Wes here.
> 
> I just took a spin through the draft and am left
> with three thoughts ...
> 
>   + You say apps don't have a good role to play even
> after citing apps
>     that 'pause' as the culprit.  I can't make these
> things agree in my
>     head.  Even in the absence of TCP-specific
> information it seems that
>     these apps could time themselves out if no
> progress is being made
>     (i.e. data getting exchanged).

Well behaved applications usually SHOULD do this, i
agree, can't be guaranteed though. Leaving the problem
to be addressed by the app, does not ensure timeliness
from the point of view that TCP resources such as
buffers and connections are scarce and by how much
etc. The scheme lacks robustness if left to the
application, and for it to be beneficial, ALL
applications running on a system should be doing it,
otherwise there is the weakest link.

> 
>     (Further, things like HTTP sends after which the
> app goes away and
>     then getting wedged into persist don't worry me
> much.  That sounds
>     like a once in a blue moon sort of situation to
> me.)
> 
>   + This seems fundamentally like an operating
> system resource issue to
>     me and not something for a networking standards
> body.  OSes deal
>     with all sorts of resource contention issues
> without standards.  Why
>     does this problem need to be solved the same way
> by each OS?

At first glance, it does seem like an OS resource
issue, but TCP connections can have data backlogged in
their send queues for a variety of reasons and for
variable amounts of time, i am not sure how a protocol
unaware OS based solution can make the distinction
between the network causing the backlog transiently,
and a legitimate vs rogue receiver causing the backlog
etc, i feel that making it TCP protocol specific
(something designed only for persist state) allows for
well controlled, granular solution(s) and heuristics
to detect specifically misbehaving receivers in this
scenario.

> 
>   + In particular, I think you have solved the
> problem wrong.  Don't get
>     me wrong ... maybe it is perfectly OK for your
> purposes and it is
>     fine with me if you want to use it.  However, I
> would want something
>     a little smarter to really avoid the mythical
> attacks you describe.
>     All you did was to cap the length of
> connections.  So, conceivably a
>     periodic attack could still keep all your
> resources busy.  It seems
>     to me that you'd want to use some sort of scheme
> that actually took
>     into account contention and resource exhaustion.
>  E.g., some sort of
>     LRU what-has-been-in-persist longest when you
> need new resources and
>     they are not available sort of culling scheme? 
> Or, when you have
>     exhausted X% of your resources to keep some in
> reserve.  That might
>     be wrong and busted itself (it is quickly off
> the top of my head),
>     but it at least takes into account the resource
> contention unlike a
>     naive timeout.
> 

The LRU scheme can be overlaid on top of the timeout
scheme to ensure a flexible and adaptive solution
quite easily, i agree. It still calls for tracking
connections bottlenecked specifically by the infinite
persistence problem.

Murali

> Just my two bits ...
> 
> allman
> 
> 
> 
> > _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www1.ietf.org/mailman/listinfo/tcpm
> 



 
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From: Mark Allman <mallman@icir.org>
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> Well behaved applications usually SHOULD do this, i
> agree, can't be guaranteed though. Leaving the problem
> to be addressed by the app, does not ensure timeliness
> from the point of view that TCP resources such as
> buffers and connections are scarce and by how much
> etc. The scheme lacks robustness if left to the
> application, and for it to be beneficial, ALL
> applications running on a system should be doing it,
> otherwise there is the weakest link.

So, let me step back.  I should have been more careful.  I think this is
a problem with the draft-not the approach.  That is, I can certainly
understand why a TCP would want to not depend on apps.  But, the text in
the i-d doesn't lead me to that conclusion and I think it is at very
best dubious in some places.

> At first glance, it does seem like an OS resource issue, but TCP
> connections can have data backlogged in their send queues for a
> variety of reasons and for variable amounts of time, i am not sure how
> a protocol unaware OS based solution can make the distinction between
> the network causing the backlog transiently, and a legitimate vs rogue
> receiver causing the backlog etc, i feel that making it TCP protocol
> specific (something designed only for persist state) allows for well
> controlled, granular solution(s) and heuristics to detect specifically
> misbehaving receivers in this scenario.

We are having a terminology problem here.  I meant "OS" as in the thing
that contains the TCP implementation.  I did not mean that an OS could
blindly reap off resources without understanding anything about TCP.

Let me put it this way ... This seems to me to be a TCP
**implementation** issue.  Why should we need to standardize this issue
and not other internal resource contention issues?  Why is this anything
but a very local problem that can be handled by the TCP implementation
in whatever way that implementation chooses?

And, then why is the proposed solution the right one?

allman




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From tcpm-bounces@ietf.org Tue Feb 20 19:27:11 2007
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From: MURALI BASHYAM <murali_bashyam@yahoo.com>
Subject: Re: [tcpm] New I-D 
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--- Mark Allman <mallman@icir.org> wrote:

> 
> > Well behaved applications usually SHOULD do this,
> i
> > agree, can't be guaranteed though. Leaving the
> problem
> > to be addressed by the app, does not ensure
> timeliness
> > from the point of view that TCP resources such as
> > buffers and connections are scarce and by how much
> > etc. The scheme lacks robustness if left to the
> > application, and for it to be beneficial, ALL
> > applications running on a system should be doing
> it,
> > otherwise there is the weakest link.
> 
> So, let me step back.  I should have been more
> careful.  I think this is
> a problem with the draft-not the approach.  That is,
> I can certainly
> understand why a TCP would want to not depend on
> apps.  But, the text in
> the i-d doesn't lead me to that conclusion and I
> think it is at very
> best dubious in some places.

If it is dubious, then we can correct it, and perhaps
we need to explain the motivation for that more
clearly.

> 
> > At first glance, it does seem like an OS resource
> issue, but TCP
> > connections can have data backlogged in their send
> queues for a
> > variety of reasons and for variable amounts of
> time, i am not sure how
> > a protocol unaware OS based solution can make the
> distinction between
> > the network causing the backlog transiently, and a
> legitimate vs rogue
> > receiver causing the backlog etc, i feel that
> making it TCP protocol
> > specific (something designed only for persist
> state) allows for well
> > controlled, granular solution(s) and heuristics to
> detect specifically
> > misbehaving receivers in this scenario.
> 
> We are having a terminology problem here.  I meant
> "OS" as in the thing
> that contains the TCP implementation.  I did not
> mean that an OS could
> blindly reap off resources without understanding
> anything about TCP.

Okay, we are on the same page :-).
> 
> Let me put it this way ... This seems to me to be a
> TCP
> **implementation** issue.  Why should we need to
> standardize this issue
> and not other internal resource contention issues? 
> Why is this anything
> but a very local problem that can be handled by the
> TCP implementation
> in whatever way that implementation chooses?

It is an implementation to correct a problem caused by
the protocol's REQUIREMENT for a sender to be in an
infinite persistence state, thus allowing for abuse by
a misbehaving receiver side application (or by a lot
of well-behaved receivers). The draft is outlining a
solution to the problem (there could be others). The
idea in the (informational) draft is to highlight to
the group that requirement of the protocol, its
consequences and a possible solution.

Are you suggesting that any requirements of the
protocol specification that lead to a potential denial
of service type of scenarios and thus causing internal
resource contention, and/or any other network issues
are all to be considered 'implementation' ? Shouldn't
we look at the requirements of the protocol a little
more carefully?

> And, then why is the proposed solution the right
> one?

It does solve the problem it sets out to based on a
"total time allowed to be in persist state" type of
cap on the duration.

Are you implying there could be other behaviours from
the receiver that may not be addressed here, such as
the receiver opening the window a little and then
closing it that fernando mentioned, thus leading to a
longer term issue? We need to build heuristics into
the solution that can detect this and other types of
anomalous receiver behaviour, and take appropriate
action. And the heuristics themselves need to be based
on a notion of a well behaved receiver in this
scenario, to avoid false positives.

Murali

> 
> allman
> 
> 
> 
> 



 
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Subject: Re: [tcpm] New I-D 
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At 08:53 p.m. 20/02/2007, Mark Allman wrote:

>Let me put it this way ... This seems to me to be a TCP
>**implementation** issue.  Why should we need to standardize this issue
>and not other internal resource contention issues?  Why is this anything
>but a very local problem that can be handled by the TCP implementation
>in whatever way that implementation chooses?

I think there are some instances of this already... e.g., Clark's RFC 
on fragment reassembly. IMHO, an Inforamtional doc on some 
implementation issues that may be troublesome might be of help.


>And, then why is the proposed solution the right one?

So far it does not look like "the right one" to me. I raised at least 
a few ways in which the attacker can fool the proposed countermeasure.


-- 
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 Feb 21 02:58:48 2007
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Subject: Re: [tcpm] New I-D posted :  draft-moncaster-tcpm-rcv-cheat-00
From: Michael Welzl <michael.welzl@uibk.ac.at>
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On Tue, 2007-02-20 at 17:27 +0000, toby.moncaster@bt.com wrote:
> All, 
> 
> As you will shortly see we have just posted a new ID called "A  TCP Test
> To Allow Senders to Identify Receiver Cheating"
> draft-moncaster-tcpm-rcv-cheat-00. This draft was written as a result of
> (..)
> http://www.cs.ucl.ac.uk/staff/B.Briscoe/pubs.html#tcp-rcv-cheat.
> 
> After some background research we decided that an actual nonce would
> probably be impractical since it would require modifications to both

If you're referring to the ECN nonce, I think that it's not fair to
compare this with it (as you also do in the draft), as it additionally
tackles a different problem: concealing the reception of an ECN mark
is much easier to do / less problematic than optimistic acking /
concealing packet loss, and this is what the ECN nonce does but your
scheme doesn't.

On a side note, the following statement in the draft:

> The proposed solution depends, like the skipped segments solution, on
> the strict requirement that all TCP receivers have to send a
> duplicate acknowledgement as soon as they receive an out-of-order
> segment.

is at odds with RFC 2581, which says:

> Out-of-order data segments SHOULD be acknowledged immediately, in
> order to accelerate loss recovery. 

...although I wonder why this is only a SHOULD, and not a MUST,
and herewith suggest to change this in 2581bis, unless there's
a good reason not to do so.

Cheers,
Michael


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From tcpm-bounces@ietf.org Wed Feb 21 05:23:45 2007
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Subject: Re: [tcpm] New I-D posted :  draft-moncaster-tcpm-rcv-cheat-00
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Hi,

On Tue, 20 Feb 2007, Rob Sherwood wrote:

> Should people decide that this two phases approach is preferable, it
> might be possible to simplify the first phase test as follows.  Rather
> then reordering segments N-1,N,N+1,..,N+D to N-1,N+1,..,N+D,N as Toby
> et. al suggest, it might be better to more frequently perform this test
> with D=1, (i.e., simply swap the order of two sequential segments) and
> after some threshold number K failures, apply the second phase randomly
> skipped segment test.

Perhaps I misunderstand, but is it not necessary to have variation in D of
at least 
        0 < D < 4
for the attack to be definitively prevented?

The "lazy ack" form of the attack just omits duplicate acks in order that a
loss is never detected.  If the above were implemented with D=1 it appears
that an attacker could always send the first duplicate ack in response to a
missing packet but omit the second or third duplicate ack.  This would seem
to allow the attack to continue but also pass the N-1,N+1,N test.

I guess if lots of packets are being lost as part of the attack this kind
of precision may be impossible on the part of the attacker so maybe the
above is not so important.

Gavin


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Subject: Re: [tcpm] New I-D 
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> It is an implementation to correct a problem caused by
> the protocol's REQUIREMENT for a sender to be in an
> infinite persistence state, thus allowing for abuse by
> a misbehaving receiver side application (or by a lot
> of well-behaved receivers). The draft is outlining a
> solution to the problem (there could be others). The
> idea in the (informational) draft is to highlight to
> the group that requirement of the protocol, its
> consequences and a possible solution.
> 
> Are you suggesting that any requirements of the
> protocol specification that lead to a potential denial
> of service type of scenarios and thus causing internal
> resource contention, and/or any other network issues
> are all to be considered 'implementation' ? Shouldn't
> we look at the requirements of the protocol a little
> more carefully?

Two things:

  + I think it is a big stretch to say this requirement should hold in
    resource constrained situations.  There are a zillion examples where
    a system is supposed to do something, but if it can't do it then it
    can't do it.  And, if a system cannot handle all the established +
    requested connections then something has to give---and, what that is
    ought to be a policy decision.

  + If you do want to read the standard as requiring a TCP to keep this
    connection in the face of everything then you don't need an
    informational because you cannot change a protocol requirement with
    an informational.

> > And, then why is the proposed solution the right
> > one?
> 
> It does solve the problem it sets out to based on a
> "total time allowed to be in persist state" type of
> cap on the duration.
> 
> Are you implying there could be other behaviours from

I am not implying anything.  In my previous note I said that a simple
timeout is a poor solution to the problem because it does not actually
take into account that resources are constrained.

Let me try an example...  Let's say we set this timer to 10 minutes.  A
TCP can stay in persist for 10 minutes.  Two possibilities:

  + The 10 minutes goes by and this is the only connection on the
    machine. 

    Why purge the connection?  The ends are both actively still
    interested and the connection is not causing any problems.

    (Clearly, we could say this is just the machine's policy---that
    connections in persist for 10 minutes get purged.  And, that is fine
    if someone wants to do that.  But, in this case the connection is
    posing no problem and so purging the connection is not addressing a
    problem, just being done due to some sort of house keeping policy.)

  + The 10 minutes goes by and the machine is heavily loaded and has
    turned away dozens or hundreds of connections because of this
    overloaded situation.

    In this case, purging the connection may offer a little relief and
    let other connections come in.  Maybe these will complete and this
    is good.  Or, maybe this is an attack and so the next connection
    will just wedge us for another 10 minutes.

    The point here is that any relief we see from this purging is by
    happenstance and not because we did something to actually address
    the resource constraint problem.  Why did we wait 10 minutes?  Why
    not 9?  Or 11?  If we want to deal with resource issues then we need
    to kick something in when resources are limited.  Kicking some
    connections out at some arbitrary point is just a *hope* that in
    some cases it will help.

What am I missing here?

allman




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From tcpm-bounces@ietf.org Wed Feb 21 10:27:41 2007
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Subject: RE: [tcpm] New I-D posted :  draft-moncaster-tcpm-rcv-cheat-00
Date: Wed, 21 Feb 2007 15:27:11 -0000
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Hi Michael.

Just to clarify, I was not talking about the ECN nonce which, as you
rightly say, is intended to address a different problem (the concealing
of ECN marks). I was actually referring to a solution presented by
Stefan Savage et al in the paper "TCP Congestion Control with a
Misbehaving Receiver". In this paper they present two versions of a
transport layer nonce which they call the "Singular Nonce" and
"Cumulative Nonce". We do look at the ECN nonce in the actual draft
since that claims to protect against the two attacks our test is
designed to identify.

Thanks for the pointer as to the slightly surprising wording in RFC2581.
As far as I can tell everyone seems to assume this was a MUST anyway...!

Toby

________________________________________________________________________
____
Toby Moncaster, <toby.moncaster@bt.com> Networks Research Centre, BT
Research
B54/70 Adastral Park, Martlesham Heath, Ipswich, IP53RE, UK.  +44 1473
648734=20


-----Original Message-----
From: Michael Welzl [mailto:michael.welzl@uibk.ac.at]=20
Sent: 21 February 2007 07:56
To: tcpm@ietf.org; Moncaster,T,Toby,CXR9 R
Subject: Re: [tcpm] New I-D posted : draft-moncaster-tcpm-rcv-cheat-00

On Tue, 2007-02-20 at 17:27 +0000, toby.moncaster@bt.com wrote:
> All,
>=20
> As you will shortly see we have just posted a new ID called "A  TCP=20
> Test To Allow Senders to Identify Receiver Cheating"
> draft-moncaster-tcpm-rcv-cheat-00. This draft was written as a result=20
> of
> (..)
> http://www.cs.ucl.ac.uk/staff/B.Briscoe/pubs.html#tcp-rcv-cheat.
>=20
> After some background research we decided that an actual nonce would=20
> probably be impractical since it would require modifications to both

If you're referring to the ECN nonce, I think that it's not fair to
compare this with it (as you also do in the draft), as it additionally
tackles a different problem: concealing the reception of an ECN mark is
much easier to do / less problematic than optimistic acking / concealing
packet loss, and this is what the ECN nonce does but your scheme
doesn't.

On a side note, the following statement in the draft:

> The proposed solution depends, like the skipped segments solution, on=20
> the strict requirement that all TCP receivers have to send a duplicate

> acknowledgement as soon as they receive an out-of-order segment.

is at odds with RFC 2581, which says:

> Out-of-order data segments SHOULD be acknowledged immediately, in=20
> order to accelerate loss recovery.

...although I wonder why this is only a SHOULD, and not a MUST, and
herewith suggest to change this in 2581bis, unless there's a good reason
not to do so.

Cheers,
Michael


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From tcpm-bounces@ietf.org Wed Feb 21 10:46:50 2007
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Subject: RE: [tcpm] New I-D posted :  draft-moncaster-tcpm-rcv-cheat-00
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Hi Rob,

We were under the impression that you were keen that the skipped
segments solution be implemented with SACK enabled as otherwise there is
a risk that it could delay the notification of congestion by up to 1 RTT
more. Apologies if we have got this wrong.

The condition we give in 6.2.4 that "Periodically and randomly, any
heavily loaded TCP sender SHOULD check the honesty of its receivers
using the probabilistic test." wasn't meant to imply that this was the
ONLY time a sender should test its receivers. It might be better in a
future draft to change it to something wider like:=20

"Any TCP sender SHOULD check the honesty of its receivers using the
probabilistic test periodically and randomly. In particular, it would be
advantageous for any sender that is heavily loaded to identify if it is
taken advantage of by an attack by a dishonest receiver".

We would hope that senders would check at other times as well, hence the
second bullet point in section 6.2.4.

Final point, as Gavin McCullagh has already pointed out, apart from
legacy Tahoe TCP implementations a receiver can be reasonably safe if it
sends less than 3 duplicate ACKs for a missing segment as this will not
trigger fast re-transmit. Consequently we have specified that the
displacement should be at least this large.

Toby

________________________________________________________________________
____
Toby Moncaster, <toby.moncaster@bt.com> Networks Research Centre, BT
Research
B54/70 Adastral Park, Martlesham Heath, Ipswich, IP53RE, UK.  +44 1473
648734=20


-----Original Message-----
From: Rob Sherwood [mailto:capveg@cs.umd.edu]=20
Sent: 20 February 2007 19:37
To: Moncaster,T,Toby,CXR9 R
Cc: tcpm@ietf.org
Subject: Re: [tcpm] New I-D posted : draft-moncaster-tcpm-rcv-cheat-00

On Tue, Feb 20, 2007 at 05:27:03PM -0000, toby.moncaster@bt.com wrote:
> All,
>=20
> As you will shortly see we have just posted a new ID called "A  TCP=20
> Test To Allow Senders to Identify Receiver Cheating"
> draft-moncaster-tcpm-rcv-cheat-00. This draft was written as a result=20
> of a request made to Bob Briscoe at IETF 67. There he mentioned to=20
> TVWG that we would be happy to produce a transport layer nonce to=20
> protect against receivers concealing lost packets. This suggestion was

> broadly welcomed by the WG and this draft is our response. Initially=20
> we intended to submit this to TSVWG as they requested it. However=20
> after checking with the co-chairs of both TCPM and TSVWG it was agreed

> that TCPM would be the best forum for the draft. The draft is=20
> available in various formats at=20
> http://www.cs.ucl.ac.uk/staff/B.Briscoe/pubs.html#tcp-rcv-cheat.

Hi Toby,

I gave a quick read through the draft on your website, and have a few
comments.


Section 5.1)=20

I think there might be some confusion here, but the skipped segment
solution that we proposed in our paper does not require the receiver
supports SACK.  Trivially, our paper presents performance results for
receivers that do and do not support SACK.  If the sender skips segment
N, any ACK > N indicates misbehavior; SACK is simply not involved for
correctness.  If SACK is not supported in the connection, then the
standard performance penalties for not implementing SACK apply, i.e.,
the receiver has to duplicate ACK each loss in turn.  Am I missing
something here?

Section 6.2.4)

"Periodically and randomly, any heavily loaded TCP sender SHOULD check
the honesty of its receivers using the probabilistic test."

I don't think it is sufficient that only heavily loaded TCP senders
perform this test.  Attackers can simply draw a small amount of traffic
from many ("jillions" to quote Mark :-) of servers, not loading any one,
but still have a large aggregate amount of traffic.  In other words, if
the solution is only applied to loaded servers, then the attack still
exists.



At a high level, I like the proposed two phase test in that it is
slightly more efficient then the randomly skipped segment test alone,
i.e., instead of honest receivers being penalized 1 RTT every test, they
are only penalized the buffer time and 1 RTT if network reordering
(pathologically?) undoes the reordering test.  It is not clear, to me at
least, how the efficiency gain weighs against the additional complexity
of code, but presumably others can comment on that.  However, my
understanding of the concern raised on this list (if I can summarize
other's opinions) was that there was a philosophical objection to
mucking up[1] the TCP code for an attack which is not being actively
exploited.
In other words, it wasn't the efficiency of the solution that was at
issue, but whether is should be applied at all.

Should people decide that this two phases approach is preferable, it
might be possible to simplify the first phase test as follows.  Rather
then reordering segments N-1,N,N+1,..,N+D to N-1,N+1,..,N+D,N as Toby
et. al suggest, it might be better to more frequently perform this test
with D=3D1, (i.e., simply swap the order of two sequential segments) and
after some threshold number K failures, apply the second phase randomly
skipped segment test.   I only suggest this as the analysis is probably
slightly simpler in that the reordering rate of pairs of packets have
been previously measured, where as the probability that the
N-1,N,N+1,..,N+D to N-1,N+1,..,N+D,N reordering would be undone is less
well studied.


- Rob
.

[1] "mucking up" is clearly a technical term :-)

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Subject: RE: [tcpm] New I-D posted :  draft-moncaster-tcpm-rcv-cheat-00
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> Just to clarify, I was not talking about the ECN nonce which, as you
> rightly say, is intended to address a different problem (the concealing
> of ECN marks). I was actually referring to a solution presented by
> Stefan Savage et al in the paper "TCP Congestion Control with a
> Misbehaving Receiver". In this paper they present two versions of a
> transport layer nonce which they call the "Singular Nonce" and
> "Cumulative Nonce". We do look at the ECN nonce in the actual draft
> since that claims to protect against the two attacks our test is
> designed to identify.

Hm, I always thought that the nonce in this paper by Stefan Savage
is the ancestor of the ECN nonce, but maybe there's a small
but significant difference there


> Thanks for the pointer as to the slightly surprising wording in RFC2581.
> As far as I can tell everyone seems to assume this was a MUST anyway...!

...and I would expect most implementations to conform with this
interpretation - so it would be interesting to learn the reasoning
behind the SHOULD. This is not the first time that I'm surprised
by something in RFC 2581, and so far, it always turned out that
there was a good reason for everything. It's just full of
such intricacies  :)

Cheers,
Michael


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Subject: RE: [tcpm] New I-D posted :  draft-moncaster-tcpm-rcv-cheat-00
Date: Wed, 21 Feb 2007 09:32:19 -0800
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Thread-Topic: [tcpm] New I-D posted :  draft-moncaster-tcpm-rcv-cheat-00
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From: "Anantha Ramaiah \(ananth\)" <ananth@cisco.com>
To: "Michael Welzl" <michael.welzl@uibk.ac.at>, <tcpm@ietf.org>,
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=20

>=20
> is at odds with RFC 2581, which says:
>=20
> > Out-of-order data segments SHOULD be acknowledged immediately, in=20
> > order to accelerate loss recovery.
>=20
> ...although I wonder why this is only a SHOULD, and not a=20
> MUST, and herewith suggest to change this in 2581bis, unless=20
> there's a good reason not to do so.

Although the RFC says SHOULD, in reality it is treated as a MUST, I
think the majority of stacks will send a ack for a out-of-order segment.
It probably makes sense to make this a MUST in RFC 2581bis if we want to
enforce some basic congestion control stuff like fast-retransmissions
(which is worded as a MUST by the way).

The only reason which I can think off why it was worded as a SHOULD
instead of a MUST is when the basic cong.control on TCP was written long
time back, "putting" many packets (even if they serve some useful
purpose) was probably considered aggressive, in the current context pure
ACK's. I am sure there are folks in the list who may have better
answers.=20

OTOH, recently, we were discussing about rate-limiting the no. of
duplicate ACK's from mis-behaving receivers in the context of RFC
2581bis, now there is a mention about
"some-receivers-that-may-exist-who-don't-send-duplicate-acks-for-out-of-
order-segments-received"=20

:-)

-Anantha

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From tcpm-bounces@ietf.org Wed Feb 21 13:34:14 2007
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From: Rob Sherwood <capveg@cs.umd.edu>
To: Gavin McCullagh <gavin.mccullagh@nuim.ie>
Subject: Re: [tcpm] New I-D posted :  draft-moncaster-tcpm-rcv-cheat-00
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On Wed, Feb 21, 2007 at 10:23:17AM +0000, Gavin McCullagh wrote:
> Hi,
> 
> On Tue, 20 Feb 2007, Rob Sherwood wrote:
> 
> > Should people decide that this two phases approach is preferable, it
> > might be possible to simplify the first phase test as follows.  Rather
> > then reordering segments N-1,N,N+1,..,N+D to N-1,N+1,..,N+D,N as Toby
> > et. al suggest, it might be better to more frequently perform this test
> > with D=1, (i.e., simply swap the order of two sequential segments) and
> > after some threshold number K failures, apply the second phase randomly
> > skipped segment test.
> 
> Perhaps I misunderstand, but is it not necessary to have variation in D of
> at least 
>         0 < D < 4
> for the attack to be definitively prevented?
> 
> The "lazy ack" form of the attack just omits duplicate acks in order that a
> loss is never detected.  If the above were implemented with D=1 it appears
> that an attacker could always send the first duplicate ack in response to a
> missing packet but omit the second or third duplicate ack.  This would seem
> to allow the attack to continue but also pass the N-1,N+1,N test.
> 
> I guess if lots of packets are being lost as part of the attack this kind
> of precision may be impossible on the part of the attacker so maybe the
> above is not so important.

Your last paragraph sums it up correctly, IMHO.  If the receiver is
actually receiving all packets, then of course it can defeat any test
-- but then there is no attack.  With Lazy ack, if the receiver were
dropping some constant percentage of packets p, then over n tests,
we can expect the receiver to fail p*n tests, and this solution can
be engineered so that k (the number of failed tests before phase 2)
is chosen to prevent the amplification rate from being too high.

In an alternate attack, the attacker could duplicate ack all packets,
but then the amplication factor suffers (1 data packet/(D+1) acks ==
1500/(D+1)*40 == 37.5x/(D+1)).   Maybe it does make sense to consider
D>1 such that the amplification factor suffers more.  Obviously, D>37,
i.e., the value the prevents *any* amplification, is too large for
practical considerations, but it does make the case for a larger value
of D (contrary to what I was suggesting earlier).

- Rob
.

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Date: Wed, 21 Feb 2007 11:13:02 -0800 (PST)
From: MURALI BASHYAM <murali_bashyam@yahoo.com>
Subject: Re: [tcpm] New I-D 
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--- Mark Allman <mallman@icir.org> wrote:

> 
> > It is an implementation to correct a problem
> caused by
> > the protocol's REQUIREMENT for a sender to be in
> an
> > infinite persistence state, thus allowing for
> abuse by
> > a misbehaving receiver side application (or by a
> lot
> > of well-behaved receivers). The draft is outlining
> a
> > solution to the problem (there could be others).
> The
> > idea in the (informational) draft is to highlight
> to
> > the group that requirement of the protocol, its
> > consequences and a possible solution.
> > 
> > Are you suggesting that any requirements of the
> > protocol specification that lead to a potential
> denial
> > of service type of scenarios and thus causing
> internal
> > resource contention, and/or any other network
> issues
> > are all to be considered 'implementation' ?
> Shouldn't
> > we look at the requirements of the protocol a
> little
> > more carefully?
> 
> Two things:
> 
>   + I think it is a big stretch to say this
> requirement should hold in
>     resource constrained situations.  There are a
> zillion examples where
>     a system is supposed to do something, but if it
> can't do it then it
>     can't do it.  And, if a system cannot handle all
> the established +
>     requested connections then something has to
> give---and, what that is
>     ought to be a policy decision.
> 
>   + If you do want to read the standard as requiring
> a TCP to keep this
>     connection in the face of everything then you
> don't need an
>     informational because you cannot change a
> protocol requirement with
>     an informational.
> 

I am not sure what the status of the draft is supposed
to be, would experimental be a more appropriate
status?
Aside from the status of the draft, the more
fundamental point is whether we agree that this is a
protocol specific requirement leading to a problem for
which there is a protocol specific/aware solution we
can put in place. 

> > > And, then why is the proposed solution the right
> > > one?
> > 
> > It does solve the problem it sets out to based on
> a
> > "total time allowed to be in persist state" type
> of
> > cap on the duration.
> > 
> > Are you implying there could be other behaviours
> from
> 
> I am not implying anything.  In my previous note I
> said that a simple
> timeout is a poor solution to the problem because it
> does not actually
> take into account that resources are constrained.
> 
> Let me try an example...  Let's say we set this
> timer to 10 minutes.  A
> TCP can stay in persist for 10 minutes.  Two
> possibilities:
> 
>   + The 10 minutes goes by and this is the only
> connection on the
>     machine. 
> 
>     Why purge the connection?  The ends are both
> actively still
>     interested and the connection is not causing any
> problems.
> 
>     (Clearly, we could say this is just the
> machine's policy---that
>     connections in persist for 10 minutes get
> purged.  And, that is fine
>     if someone wants to do that.  But, in this case
> the connection is
>     posing no problem and so purging the connection
> is not addressing a
>     problem, just being done due to some sort of
> house keeping policy.)
> 
>   + The 10 minutes goes by and the machine is
> heavily loaded and has
>     turned away dozens or hundreds of connections
> because of this
>     overloaded situation.
> 
>     In this case, purging the connection may offer a
> little relief and
>     let other connections come in.  Maybe these will
> complete and this
>     is good.  Or, maybe this is an attack and so the
> next connection
>     will just wedge us for another 10 minutes.
> 
>     The point here is that any relief we see from
> this purging is by
>     happenstance and not because we did something to
> actually address
>     the resource constraint problem.  Why did we
> wait 10 minutes?  Why
>     not 9?  Or 11?  If we want to deal with resource
> issues then we need
>     to kick something in when resources are limited.
>  Kicking some
>     connections out at some arbitrary point is just
> a *hope* that in
>     some cases it will help.
> 
> What am I missing here?

I thought we agreed that a resource based demand
driven scheme (based on who has been idle for the
longest time) can be used on top of the timeout based
policy scheme. Doesn't that address the issue are
raising here?

Thx,
Murali
> 
> allman
> 
> 
> 
> 



 
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From: "Caitlin Bestler" <caitlinb@broadcom.com>
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I am very skeptical that there is a transport layer problem here,
as opposed to an application layer problem.

Basically, I do not accept the following premise:

	A large server relies on honest network congestion feedback to
	efficiently apportion its own resources between receivers. If
	such a large server devotes an excessive fraction of its own
	resources to misbehaving receivers, it may well hit its own
	resource limits and have to starve other half-connections even
	if their network path has spare capacity.

I would rather contend:

	A large server does not rely on network congestion feedback
alone
	to efficiently apportion its own resources between receivers. If
	such a large receiver devotes an excessive fraction of its own
	resources to any receiver (whether misbehaving, malicious or=20
	merely greedy) it may well hit its own resource limits and
	have to starve other half-connections even if their network
	path has spare capacity.

A server application that is willing to transmit multi-megabit flows
to an anonymous client based solely on the alleged receive capacity
of the receiver is so gullible it almost qualifies as a "willing
accomplice".

My understanding of this problem assumes the following:

	DoS attacks that seek to cause a server to send very large
streams,
	and thereby cause local congestion at the server end, require
establishing
	actual connections. Therefore the source of the attack is
traceable.
	An attacker is unlikely to use their own machine for a traceable
attack.
	Therefore these attacks would be launched from compromised
machines, from
	a botnet.

	A compromised machine can already be used to "honestly" receive
a stream
	at the compromised machine's capacity rate. The installed
malware can
	simply throw the traffic away with very low CPU overhead. If
clever, it
	can limit the rate to a fraction of the local bandwidth to avoid
arousing
	suspicion of the machine's legitimate owner.

	Therefore the proposed test is intended as a countermeasure
against attackers
	seeking to use optimistic acking to get a multiplier effect. By
committing
	virtually all of the local machines receive bandwidth it can get
the server
	to commit several times that on the server's local network.

	The fake receiver will ack packets that are actually reeived as
though all
	prior packets had been received. An optimal attacker would
probably simulate
	a long NAPI poll cycle of 100 HZ and ack the highest TCP
sequence received
	ignoring all gaps. It would thereby simulate the behavior of an
honest
	receiver that consolidated ACKs on a 100 HZ collection cycle.

An important constraint is that the attacker wants to generate no fewer
acks than
an honest receiver actually receiving the packets would have. The
greatest ACK to
receive ratio that I can think of, this is also legitimate, occurs when
an honest
receiver generates a single ACK for all traffic received in a collection
period.

A receiver that is generating ACKs less frequently than every other sent
packet
or 100HZ AND is claiming that the connection is experiencing no
congestion is
inherently suspicious. Even if it manages to be RFC compliant, the
sender is
under no obligation to continue sending traffic to this client at higher
and
higher rates.

This is especially true since even with the proposed test, an attacker
can=20
always recruit compromised attackers to act as a bit bucket drain at
typical
home broadband rates. Since the goal of optimistic acking is to get a
multiplier
effect, the rates in question would be several times higher than typical
home
broadband capacities.

So wouldn't having the application layer merely cap individual flows to
anonymous
clients be just as effective a counter-measure as fiddling with TCP?

And wouldn't it be less harmful?

The proposed test *is* harmful. It artifically increases the amount of
packet
re-ordering in the network. Receivers may already have engineered
resources
to provide resilient responses to re-ordered packets. Artificially
caused
packet reordering as a routine test eats away at those safety margins.
And
the duplicate acks generated ARE excess traffic.

As an *administrative* test done when a flow already appears suspicious
there would be no real problem. But as a routine test to determine when
a flow is suspicious there is an inherent problem. Either the test is
performed too seldom to be effective, or it is performed frequently
enough to have a measurable impact on the performance of at least
some receivers.




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From tcpm-bounces@ietf.org Wed Feb 21 15:48:25 2007
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MURALI BASHYAM wrote:

>=20
> I am not sure what the status of the draft is supposed to be,
> would experimental be a more appropriate status?
> Aside from the status of the draft, the more fundamental
> point is whether we agree that this is a protocol specific
> requirement leading to a problem for which there is a
> protocol specific/aware solution we can put in place.
>=20

I have always assumed the following:

	Any application MAY disconnect from any peer when
	it no longer finds maintaining the connection with
	the peer to be of value.

	An application SHOULD disconnect from any peer when
	continued support of that peer will have an adverse
	impact on other connections, especially when the
	peer is acting suspiciously.

Therefore any application MAY and perhaps SHOULD disconnect
from a suspiciously idle peer, especially before maintaining
the idle connection adversely impacts active connections.

Are you suggesting that there is anything in any protocol
requirement that contradicts this?


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From tcpm-bounces@ietf.org Wed Feb 21 17:34:36 2007
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From: Rob Sherwood <capveg@cs.umd.edu>
To: toby.moncaster@bt.com
Subject: Re: [tcpm] New I-D posted :  draft-moncaster-tcpm-rcv-cheat-00
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On Wed, Feb 21, 2007 at 03:46:10PM -0000, toby.moncaster@bt.com wrote:
> Hi Rob,
> 
> We were under the impression that you were keen that the skipped
> segments solution be implemented with SACK enabled as otherwise there is
> a risk that it could delay the notification of congestion by up to 1 RTT
> more. Apologies if we have got this wrong.

I'm not sure that I see that risk.  I think both this point and
the point below rely on the same aspect of TCP: when an out of order
segment arrives, the receiver immediately sends a duplicate ACK for the
previously acknowledged segment.  Thus, as long as the receiver sends
a single duplicate ack for the skipped segment, then the sender know
it is honest.  Because there is no delay in the dup ack generation,
the penalty for receiving the skipped segment is at most 1 RTT.


> "Any TCP sender SHOULD check the honesty of its receivers using the
> probabilistic test periodically and randomly. In particular, it would be
> advantageous for any sender that is heavily loaded to identify if it is
> taken advantage of by an attack by a dishonest receiver".

This wording is definitely clearer.

> Final point, as Gavin McCullagh has already pointed out, apart from
> legacy Tahoe TCP implementations a receiver can be reasonably safe if it
> sends less than 3 duplicate ACKs for a missing segment as this will not
> trigger fast re-transmit. Consequently we have specified that the
> displacement should be at least this large.

Maybe I'm the one who is confused, but as I understand it, a single
duplicate ACK (for the correct segment) is enough to differentiate
an honest receiver from a malicious one, so why do you need multiple
duplicate acks?   Fast retransmit should never be an issue, because 
the sender should not perform congestion control in response to a
malicious receiver test.  

- Rob
.

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From tcpm-bounces@ietf.org Wed Feb 21 17:38:07 2007
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I think I agree with Wes, David and Mark that this is more of an OS than 
a protocol issue (if I'm characterizing their comments correctly), and 
Fernando that the approach is not powerful enough to solve the broader 
problem when viewed as an attack.

To expand a bit, I had a hallway conversation a couple years ago with 
Stanislav Shalunov about his netkill [1] attack, and more generally 
about contention for send buffer memory.  (I think the problem you're 
addressing falls precisely into this broader category.)  What I took 
away from this (Stas may or may not agree) is that since the resource 
contention occurs at the operating system level (competing for scarce 
kernel memory), that is the right place to arbitrate the contention. 
Each host may want to apply a different policy based on what it 
considers important.

As proof of concept, I wrote up a simple script (I think only about 30 
lines of python) that used the TCP ESTATS MIB to monitor connections, 
and kill off ones making no progress while consuming the most memory. 
This proved an effective defense against netkill, and I think would 
solve the problem described in the draft as well.  It would be easy 
enough to write a kernel thread to do the same if the system doesn't do 
ESTATS.

   -John


[1] http://shlang.com/netkill/

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From tcpm-bounces@ietf.org Wed Feb 21 18:20:57 2007
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Anantha Ramaiah (ananth) wrote:
>  
> 
>> is at odds with RFC 2581, which says:
>>
>>> Out-of-order data segments SHOULD be acknowledged immediately, in 
>>> order to accelerate loss recovery.
>> ...although I wonder why this is only a SHOULD, and not a 
>> MUST, and herewith suggest to change this in 2581bis, unless 
>> there's a good reason not to do so.
> 
> Although the RFC says SHOULD, in reality it is treated as a MUST, I
> think the majority of stacks will send a ack for a out-of-order segment.
> It probably makes sense to make this a MUST in RFC 2581bis if we want to
> enforce some basic congestion control stuff like fast-retransmissions
> (which is worded as a MUST by the way).
> 
> The only reason which I can think off why it was worded as a SHOULD
> instead of a MUST is when the basic cong.control on TCP was written long
> time back, "putting" many packets (even if they serve some useful
> purpose) was probably considered aggressive, in the current context pure
> ACK's. I am sure there are folks in the list who may have better
> answers. 
> 
> OTOH, recently, we were discussing about rate-limiting the no. of
> duplicate ACK's from mis-behaving receivers in the context of RFC
> 2581bis, now there is a mention about
> "some-receivers-that-may-exist-who-don't-send-duplicate-acks-for-out-of-
> order-segments-received" 


I would be reluctant to change this to a MUST.  A few things that come 
to mind:


As you mention, some receivers might want to adopt a slightly different 
ACK strategy if there's persistent reordering, to avoid having to ACK 
every packet in steady state, and to prevent triggering fast retransmit 
on senders that don't do NCR (RFC4653) or similar.

In CPU- or interrupt-bound senders, suddenly doubling the rate of ACKs 
it must process can have adverse effects on performance.  It might make 
sense for some high-performance flows with large windows to do delayed 
ACK all the time.  (With large windows and a low loss rate, there's 
little risk.)

Paths with severe asymmetry may want to thin ACKs.  There were some 
papers on this in the mid to late 90's coming out of work with 
satellites and ADSL.

There's been some talk lately about reverse path congestion control. 
(This would be much harder with TCP than DCCP, but maybe someone can 
figure it out?)


Some of this stuff might be hair-brained or not, but we don't want to 
restrict flexibility unless it's really necessary.

   -John

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=20

> -----Original Message-----
> From: John Heffner [mailto:jheffner@psc.edu]=20
> Sent: Wednesday, February 21, 2007 3:20 PM
> To: tcpm@ietf.org
> Cc: Anantha Ramaiah (ananth); Michael Welzl; toby.moncaster@bt.com
> Subject: Re: [tcpm] New I-D posted : draft-moncaster-tcpm-rcv-cheat-00
>=20
> Anantha Ramaiah (ananth) wrote:
> > =20
> >=20
> >> is at odds with RFC 2581, which says:
> >>
> >>> Out-of-order data segments SHOULD be acknowledged immediately, in=20
> >>> order to accelerate loss recovery.
> >> ...although I wonder why this is only a SHOULD, and not a=20
> MUST, and=20
> >> herewith suggest to change this in 2581bis, unless there's a good=20
> >> reason not to do so.
> >=20
> > Although the RFC says SHOULD, in reality it is treated as a MUST, I=20
> > think the majority of stacks will send a ack for a=20
> out-of-order segment.
> > It probably makes sense to make this a MUST in RFC 2581bis=20
> if we want=20
> > to enforce some basic congestion control stuff like=20
> > fast-retransmissions (which is worded as a MUST by the way).
> >=20
> > The only reason which I can think off why it was worded as a SHOULD=20
> > instead of a MUST is when the basic cong.control on TCP was written=20
> > long time back, "putting" many packets (even if they serve=20
> some useful
> > purpose) was probably considered aggressive, in the current context=20
> > pure ACK's. I am sure there are folks in the list who may=20
> have better=20
> > answers.
> >=20
> > OTOH, recently, we were discussing about rate-limiting the no. of=20
> > duplicate ACK's from mis-behaving receivers in the context of RFC=20
> > 2581bis, now there is a mention about
> >=20
> "some-receivers-that-may-exist-who-don't-send-duplicate-acks-for-out-o
> > f-
> > order-segments-received"=20
>=20
>=20
> I would be reluctant to change this to a MUST.  A few things=20
> that come to mind:
>=20
>=20
> As you mention, some receivers might want to adopt a slightly=20
> different=20
> ACK strategy if there's persistent reordering, to avoid having to ACK=20
> every packet in steady state, and to prevent triggering fast=20
> retransmit=20
> on senders that don't do NCR (RFC4653) or similar.

I am not aware of receivers that rate limit the ACK's due to the fact
that there was persistent re-ordering in the network. YMMV.

>=20
> In CPU- or interrupt-bound senders, suddenly doubling the=20
> rate of ACKs=20
> it must process can have adverse effects on performance.  It

Well, I would classify this as the problem of the receiver which isn't
able to process these ACK's. So, if I understand what you are saying
correctly, if the peer to that receiver isn't sending a duplicate ACK
for every out-of-order packet received, things would work well in this
particular case, in cases if it peers with a box (host/router any TCP
device) which follws the SHOULD clause seriously and ack's every other
out-of-order-packet, then the receiver is in trouble? After all the
receiver can chose to drop the ACK, when it detects conditions that
effect performance. It is not rocket-sicence to implement performance
monitors and take corrective actions.

<snip>

>=20
> Paths with severe asymmetry may want to thin ACKs.  There were some=20
> papers on this in the mid to late 90's coming out of work with=20
> satellites and ADSL.

Ok.

>=20
> There's been some talk lately about reverse path congestion control.=20
> (This would be much harder with TCP than DCCP, but maybe someone can=20
> figure it out?)

By reverse path, you mean ACK congestion control? Data transfer can be
bi-directional.

>=20
>=20
> Some of this stuff might be hair-brained or not, but we don't want to=20
> restrict flexibility unless it's really necessary.

I agree. My point ( I haven't surveyed the whole internet/intranets
obviously) is :-

- since it is believed that most of the stacks existing today, ACKs
every out-of-order segment it received, "SHOULD" clause has been in the
experimental stage for a while..

- The internet has changed a long way what it was 20+ years back when
the original TCP specs were written and some of the assumptions that
were made are no longer commonplace. In other words the assumptions
which were common has become rare today.

Hence my thinking ( I know some folks wouldn't like it) is we don't lose
much by changing it. May be we don't lose much by not changing ( except
that we would be in a situation of not refelecting the common practice
in the standards) as well :-)

-Anantha

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From tcpm-bounces@ietf.org Thu Feb 22 02:28:27 2007
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Subject: RE: [tcpm] New I-D posted :  draft-moncaster-tcpm-rcv-cheat-00
From: Michael Welzl <michael.welzl@uibk.ac.at>
To: "Anantha Ramaiah (ananth)" <ananth@cisco.com>
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> > There's been some talk lately about reverse path congestion control. 
> > (This would be much harder with TCP than DCCP, but maybe someone can 
> > figure it out?)
> 
> By reverse path, you mean ACK congestion control? Data transfer can be
> bi-directional.

Funny enough, I stumbled over this SHOULD when thinking about ACK
congestion control, where it could be useful for the sender to be
able to count the number of ACKs that the receiver would normally
have to send during a loss period.

In any case, it's the concern of a new mechanism, yet to be specified,
how to do ACK congestion control, and not the concern of RFC 2581 -
using a SHOULD instead of a MUST for sending ACKs to out-of-order
segments doesn't implement or even help ACK congestion control,
but it actually seems to be a hindrance.

Cheers,
Michael


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From tcpm-bounces@ietf.org Thu Feb 22 11:18:52 2007
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Subject: RE: [tcpm] New I-D posted :  draft-moncaster-tcpm-rcv-cheat-00
Date: Thu, 22 Feb 2007 16:17:37 -0000
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Hi Rob,

The rationale behind our mechanism is that the latency in receiving the
skipped segment matters far less than any latency caused in the
potential signalling of a congestion event occurring during the honesty
test. This aspect is never significant with TCP SACK because there can
be enough information in a single acknowledgment to signal both that a
segment is missing, and that a later segment has experienced congestion.

The mechanism you have presented in [Sherwood05] defers sending the
skipped segment until TCP-SACK or a dup ack signals it is missing ,
which will take about one RTT. It will take another RTT for the skipped
segment to be duly acknowledged.

If SACK is not used, the problem is that the receiver cannot signal any
congestion event for about one RTT, from the time when it detects the
skipped segment is missing until the time it receives it. TCP can only
advance the acknowledgement number when it receives in-order data
(strictly data which advances the receive window). In other words your
test will generate a stream of dup acks all for the segment before the
skipped segment until such time as the skipped segment is actually
received. =20

The advantage of the mechanism in [draft-moncaster-tcpm-rcv-cheat-00] is
that it reduces the time it takes for the source to send the skipped
segment to a (random) fraction of the congestion window, which on
average will be less than half an RTT.

Forcing a receiver to send a single duplicate ack does partially show
their honesty in that it makes it hard for them to send optimistic acks.
However it still allows them to conceal actual segment losses as they
will generally not suffer any penalty (fast retransmit) until they have
sent three duplicate acknowledgements.

Toby

-----Original Message-----
From: Rob Sherwood [mailto:capveg@cs.umd.edu]=20
Sent: 21 February 2007 22:30
To: Moncaster,T,Toby,CXR9 R
Cc: capveg@cs.umd.edu; tcpm@ietf.org
Subject: Re: [tcpm] New I-D posted : draft-moncaster-tcpm-rcv-cheat-00

On Wed, Feb 21, 2007 at 03:46:10PM -0000, toby.moncaster@bt.com wrote:
> Hi Rob,
>=20
> We were under the impression that you were keen that the skipped=20
> segments solution be implemented with SACK enabled as otherwise there=20
> is a risk that it could delay the notification of congestion by up to=20
> 1 RTT more. Apologies if we have got this wrong.

I'm not sure that I see that risk.  I think both this point and the
point below rely on the same aspect of TCP: when an out of order segment
arrives, the receiver immediately sends a duplicate ACK for the
previously acknowledged segment.  Thus, as long as the receiver sends a
single duplicate ack for the skipped segment, then the sender know it is
honest.  Because there is no delay in the dup ack generation, the
penalty for receiving the skipped segment is at most 1 RTT.


> "Any TCP sender SHOULD check the honesty of its receivers using the=20
> probabilistic test periodically and randomly. In particular, it would=20
> be advantageous for any sender that is heavily loaded to identify if=20
> it is taken advantage of by an attack by a dishonest receiver".

This wording is definitely clearer.

> Final point, as Gavin McCullagh has already pointed out, apart from=20
> legacy Tahoe TCP implementations a receiver can be reasonably safe if=20
> it sends less than 3 duplicate ACKs for a missing segment as this will

> not trigger fast re-transmit. Consequently we have specified that the=20
> displacement should be at least this large.

Maybe I'm the one who is confused, but as I understand it, a single
duplicate ACK (for the correct segment) is enough to differentiate an
honest receiver from a malicious one, so why do you need multiple
duplicate acks?   Fast retransmit should never be an issue, because=20
the sender should not perform congestion control in response to a
malicious receiver test. =20

- Rob
.

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From tcpm-bounces@ietf.org Thu Feb 22 14:25:57 2007
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Michael Welzl wrote:
>>> There's been some talk lately about reverse path congestion control. 
>>> (This would be much harder with TCP than DCCP, but maybe someone can 
>>> figure it out?)
>> By reverse path, you mean ACK congestion control? Data transfer can be
>> bi-directional.
> 
> Funny enough, I stumbled over this SHOULD when thinking about ACK
> congestion control, where it could be useful for the sender to be
> able to count the number of ACKs that the receiver would normally
> have to send during a loss period.
> 
> In any case, it's the concern of a new mechanism, yet to be specified,
> how to do ACK congestion control, and not the concern of RFC 2581 -
> using a SHOULD instead of a MUST for sending ACKs to out-of-order
> segments doesn't implement or even help ACK congestion control,
> but it actually seems to be a hindrance.

On the other hand, I don't know how you could possibly hope to do ACK 
congestion control when congestion on the forward path MUST cause you to 
suddenly double your rate of sending ACKs.

This also leads to another possible reason why you might want to 
continue delayed ACK during reordering/loss: on half-duplex links, 
especially wireless which may have a high per-packet cost, doubling your 
ACK rate in response to congestion actually makes the congestion worse.

I don't really want to get into defending specifics of any of these 
general ideas, but I'm just trying to say I wouldn't want to entirely 
rule out any of these ideas from future TCP, unless a strong case is 
made to do so.

   -John

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From tcpm-bounces@ietf.org Thu Feb 22 16:01:38 2007
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To: "John Heffner" <jheffner@psc.edu>
Subject: AW: [tcpm] New I-D posted :  draft-moncaster-tcpm-rcv-cheat-00
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> > In any case, it's the concern of a new mechanism, yet to be specified,
> > how to do ACK congestion control, and not the concern of RFC 2581 -
> > using a SHOULD instead of a MUST for sending ACKs to out-of-order
> > segments doesn't implement or even help ACK congestion control,
> > but it actually seems to be a hindrance.
> 
> On the other hand, I don't know how you could possibly hope to do ACK 
> congestion control when congestion on the forward path MUST cause you to 
> suddenly double your rate of sending ACKs.

What I meant with "new mechanism, yet to be specified" is that
we could, at some point, decide to specify an ACK congestion
control mechanism which would, under controlled conditions,
reduce the ACK rate, and override this MUST with something
else (i.e. update RFC 2581bis) - but since such a mechanism
does not (yet) exist, 2581bis isn't the right place for this.


> This also leads to another possible reason why you might want to 
> continue delayed ACK during reordering/loss: on half-duplex links, 
> especially wireless which may have a high per-packet cost, doubling your 
> ACK rate in response to congestion actually makes the congestion worse.
> 
> I don't really want to get into defending specifics of any of these 
> general ideas, but I'm just trying to say I wouldn't want to entirely 
> rule out any of these ideas from future TCP, unless a strong case is 
> made to do so.

It wouldn't rule this out from future RFCs on TCP, and TCP
implementations that are based on them. It merely rules it
out of future TCP implementations (and here, I would argue
that a larger step is needed for proper ACK congestion
control than the flexibility that is granted by the SHOULD
in RFC 2581).

Cheers,
Michael

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Cc: tcpm@ietf.org, "Anantha Ramaiah \(ananth\)" <ananth@cisco.com>
Subject: [tcpm] Re: RFC2581bis ooo acking WAS: New I-D posted :
	draft-moncaster-tcpm-rcv-cheat-00
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I guess I'm not quite sure what you mean (maybe we are using language 
that's too indirect).

To get to the heart of the matter: do you advocate changing the first 
SHOULD in Section 3.2 of RFC2581 to a MUST in 2581bis?

This is not a trivial change.

Thanks,
   -John


Michael Welzl wrote:
>>> In any case, it's the concern of a new mechanism, yet to be specified,
>>> how to do ACK congestion control, and not the concern of RFC 2581 -
>>> using a SHOULD instead of a MUST for sending ACKs to out-of-order
>>> segments doesn't implement or even help ACK congestion control,
>>> but it actually seems to be a hindrance.
>> On the other hand, I don't know how you could possibly hope to do ACK 
>> congestion control when congestion on the forward path MUST cause you to 
>> suddenly double your rate of sending ACKs.
> 
> What I meant with "new mechanism, yet to be specified" is that
> we could, at some point, decide to specify an ACK congestion
> control mechanism which would, under controlled conditions,
> reduce the ACK rate, and override this MUST with something
> else (i.e. update RFC 2581bis) - but since such a mechanism
> does not (yet) exist, 2581bis isn't the right place for this.
> 
> 
>> This also leads to another possible reason why you might want to 
>> continue delayed ACK during reordering/loss: on half-duplex links, 
>> especially wireless which may have a high per-packet cost, doubling your 
>> ACK rate in response to congestion actually makes the congestion worse.
>>
>> I don't really want to get into defending specifics of any of these 
>> general ideas, but I'm just trying to say I wouldn't want to entirely 
>> rule out any of these ideas from future TCP, unless a strong case is 
>> made to do so.
> 
> It wouldn't rule this out from future RFCs on TCP, and TCP
> implementations that are based on them. It merely rules it
> out of future TCP implementations (and here, I would argue
> that a larger step is needed for proper ACK congestion
> control than the flexibility that is granted by the SHOULD
> in RFC 2581).
> 
> Cheers,
> Michael


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Cc: tcpm@ietf.org, "Anantha Ramaiah \(ananth\)" <ananth@cisco.com>
Subject: [tcpm] Re: RFC2581bis ooo acking WAS: New I-D posted :
	draft-moncaster-tcpm-rcv-cheat-00
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I guess I'm not quite sure what you mean (maybe we are using language 
that's too indirect).

To get to the heart of the matter: do you advocate changing the first 
SHOULD in Section 3.2 of RFC2581 to a MUST in 2581bis?

This is not a trivial change.

Thanks,
   -John


Michael Welzl wrote:
>>> In any case, it's the concern of a new mechanism, yet to be specified,
>>> how to do ACK congestion control, and not the concern of RFC 2581 -
>>> using a SHOULD instead of a MUST for sending ACKs to out-of-order
>>> segments doesn't implement or even help ACK congestion control,
>>> but it actually seems to be a hindrance.
>> On the other hand, I don't know how you could possibly hope to do ACK 
>> congestion control when congestion on the forward path MUST cause you to 
>> suddenly double your rate of sending ACKs.
> 
> What I meant with "new mechanism, yet to be specified" is that
> we could, at some point, decide to specify an ACK congestion
> control mechanism which would, under controlled conditions,
> reduce the ACK rate, and override this MUST with something
> else (i.e. update RFC 2581bis) - but since such a mechanism
> does not (yet) exist, 2581bis isn't the right place for this.
> 
> 
>> This also leads to another possible reason why you might want to 
>> continue delayed ACK during reordering/loss: on half-duplex links, 
>> especially wireless which may have a high per-packet cost, doubling your 
>> ACK rate in response to congestion actually makes the congestion worse.
>>
>> I don't really want to get into defending specifics of any of these 
>> general ideas, but I'm just trying to say I wouldn't want to entirely 
>> rule out any of these ideas from future TCP, unless a strong case is 
>> made to do so.
> 
> It wouldn't rule this out from future RFCs on TCP, and TCP
> implementations that are based on them. It merely rules it
> out of future TCP implementations (and here, I would argue
> that a larger step is needed for proper ACK congestion
> control than the flexibility that is granted by the SHOULD
> in RFC 2581).
> 
> Cheers,
> Michael


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Subject: [tcpm] AW: RFC2581bis ooo acking WAS: New I-D posted :
	draft-moncaster-tcpm-rcv-cheat-00
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> To get to the heart of the matter: do you advocate changing the first 
> SHOULD in Section 3.2 of RFC2581 to a MUST in 2581bis?

Yes,
 
> This is not a trivial change.

but I think it's a good one (and I don't see why it would
be *that* problematic).

Cheers,
Michael

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





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From tcpm-bounces@ietf.org Thu Feb 22 16:25:06 2007
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Subject: [tcpm] AW: RFC2581bis ooo acking WAS: New I-D posted :
	draft-moncaster-tcpm-rcv-cheat-00
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> To get to the heart of the matter: do you advocate changing the first 
> SHOULD in Section 3.2 of RFC2581 to a MUST in 2581bis?

Yes,
 
> This is not a trivial change.

but I think it's a good one (and I don't see why it would
be *that* problematic).

Cheers,
Michael

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From tcpm-bounces@ietf.org Fri Feb 23 15:54:02 2007
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts 
directories.
This draft is a work item of the TCP Maintenance and Minor Extensions Working Group of the IETF.

	Title		: Improving TCP's Robustness to Blind In-Window Attacks
	Author(s)	: A. Ramaiah, et al.
	Filename	: draft-ietf-tcpm-tcpsecure-07.txt
	Pages		: 26
	Date		: 2007-2-23
	
TCP has historically been considered protected against spoofed packet
   injection attacks by relying on the fact that it is difficult to
   guess the 4-tuple (the source and destination IP addresses and the
   source and destination ports) in combination with the 32 bit sequence
   number(s).  A combination of increasing window sizes and applications
   using a longer term connections (e.g.  H-323 or Border Gateway
   Protocol [RFC4271]) have left modern TCP implementation more
   vulnerable to these types of spoofed packet injection attacks.

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

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

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On Thu, Feb 22, 2007 at 10:23:19PM +0100, Michael Welzl wrote:
> > To get to the heart of the matter: do you advocate changing the first=
=20
> > SHOULD in Section 3.2 of RFC2581 to a MUST in 2581bis?
>=20
> Yes,
> =20
> > This is not a trivial change.
>=20
> but I think it's a good one (and I don't see why it would
> be *that* problematic).

My thinking is that the SHOULD in question should reamin as it stands.
The advantages of the receiver behavior are obvious, but it's also
perfectly clear that this behavior is not imperative to the correct
function of the TCP protocol, nor will disobeying the SHOULD impair
other users of the network (the source will enter fast retransmit
somewhat later than it optimally could have, but there are plenty of
other possible causes of that delay).

I don't think anyone argues that a good implementation will send the
ACKs.  I don't understand the argument that the system must be
constrained to do it or be non-conformant.  Optimizations are at most
SHOULDs almost by definition.

That's all IMHO, w/o a chair hat.

--=20
Ted Faber
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Subject: Re: [tcpm] AW: RFC2581bis ooo acking WAS: New I-D posted :
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On Fri, Feb 23, 2007 at 03:14:29PM -0800, Ted Faber wrote:
> I don't think anyone argues that a good implementation will send the
> ACKs.=20

Strike that; reverse it.

We agree a good implementation will send the ACKs.

--=20
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Subject: RE: [tcpm] AW: RFC2581bis ooo acking WAS: New I-D posted :
	draft-moncaster-tcpm-rcv-cheat-00
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Ted Faber wrote:
> On Fri, Feb 23, 2007 at 03:14:29PM -0800, Ted Faber wrote:
>> I don't think anyone argues that a good implementation will send the
>> ACKs.
>=20
> Strike that; reverse it.
>=20
> We agree a good implementation will send the ACKs.

I agree with that as long as "send" is understood to=20
be inclusive of piggy-backing the "send" on a TCP
segment that is already scheduled for transmit.

That is, when the draft says "immediately send" it
means "do not delay, in fact use a pure ack if
necessary."


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On Fri, Feb 23, 2007 at 03:59:33PM -0800, Caitlin Bestler wrote:
> Ted Faber wrote:
> > On Fri, Feb 23, 2007 at 03:14:29PM -0800, Ted Faber wrote:
> >> I don't think anyone argues that a good implementation will send the
> >> ACKs.
> >=20
> > Strike that; reverse it.
> >=20
> > We agree a good implementation will send the ACKs.
>=20
> I agree with that as long as "send" is understood to=20
> be inclusive of piggy-backing the "send" on a TCP
> segment that is already scheduled for transmit.
>=20
> That is, when the draft says "immediately send" it
> means "do not delay, in fact use a pure ack if
> necessary."

We agree.

It's my interpretation that TCP should always piggyback an ACK when
possible.  That's how I read P. 74 of RFC 793, anyway:=20

      ESTABLISHED STATE
      FIN-WAIT-1 STATE
      FIN-WAIT-2 STATE

	Once in the ESTABLISHED state, it is possible to deliver segment
	text to user RECEIVE buffers.   [...]

	When the TCP takes responsibility for delivering the data to the
	user it must also acknowledge the receipt of the data.

	[...]

	Send an acknowledgment of the form:

          <SEQ=3DSND.NXT><ACK=3DRCV.NXT><CTL=3DACK>

	This acknowledgment should be piggybacked on a segment being
	transmitted if possible without incurring undue delay.

To my eyes that's a SHOULD in the last paragraph, but (loosely) because=20
793 < 2119 that typography wasn't in use.


--=20
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The following is a summary of feedback from the last call:

- Wes Eddy endorsed the doc

- Alfred Hoenes provided detailed input

	- numerous typos were addressed

	- sec 2.4 had an error, where the attack takes only 20 seconds
	for a 1Mbps link; this description was updated to indicate that
	such an attack is less likely primarily because of port
	randomization, and now cites Larsen/Gont's recent I-D

	- sec 5.2 AH was corrected from "deprecated" to
	"being phased out", and now cites 4301

	- requested addressing NATs
	a paragraph was appended to the security considerations section

- Lars Eggert provided input

	- also noticed the sec 2.4 error above

- Pekka Savola provided input

	- sec 3.2.1, paragraph 3 was reworded according to his
	suggested text on filtering

	- draft-ietf-rtgwg-rfc3682bis-09.txt was added to the
	references where BITW was discussed

- the document was updated with revised BCP 78 boilerplate

All the above modifications were confirmed as sufficient by the parties
above. The revised document, -06, has already been submitted to the I-D
repository (and passed IDnits). It should be available shortly; until
then, it is available at:
http://www.isi.edu/touch/pubs/draft-ietf-tcpm-tcp-antispoof-06.txt

Joe


Ted Faber wrote:
> On Wed, Jan 17, 2007 at 05:24:41PM -0800, Ted Faber wrote:
>> 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.tx=
t
>>
>> Comments (including positive ones, please) to the list.
>>
>> This WGLC will end 2 Feb 2007 - a little more than 2 weeks from now.
>=20
> 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 th=
e
> 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
>=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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Date: Sat, 24 Feb 2007 09:19:38 +0100
From: Michael Welzl <Michael.Welzl@uibk.ac.at>
To: Ted Faber <faber@ISI.EDU>
Subject: Re: [tcpm] AW: RFC2581bis ooo acking WAS: New I-D posted :
	draft-moncaster-tcpm-rcv-cheat-00
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> My thinking is that the SHOULD in question should reamin as it stands.
> The advantages of the receiver behavior are obvious, but it's also
> perfectly clear that this behavior is not imperative to the correct
> function of the TCP protocol, nor will disobeying the SHOULD impair
> other users of the network (the source will enter fast retransmit
> somewhat later than it optimally could have, but there are plenty of
> other possible causes of that delay).
>
> I don't think anyone argues that a good implementation will send the
> ACKs.  I don't understand the argument that the system must be
> constrained to do it or be non-conformant.  Optimizations are at most
> SHOULDs almost by definition.

What confuses me is this discrepance that, on the one hand,
a lot of effort was made to avoid entering slow start by
reasonably using DupACKs (NewReno, the SACK scoreboard)
and getting enough of them (Limited Transmit), while on
the other, we simply leave it up to the receiver to send
less than it should.

Does it make sense to allow a receiver to do something which
has no obvious benefit, but clearly leads to undesirable
behavior?

Cheers,
Michael

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From: Ethan Blanton <eblanton@cs.ohiou.edu>
To: tcpm@ietf.org
Subject: Re: [tcpm] AW: RFC2581bis ooo acking WAS: New I-D posted :
	draft-moncaster-tcpm-rcv-cheat-00
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[The "something" below being delaying a dup ACK for a segment received
 out-of-order using delayed ACKs.]

Michael Welzl spake unto us the following wisdom:
> Does it make sense to allow a receiver to do something which
> has no obvious benefit, but clearly leads to undesirable
> behavior?

Yes, if it does not break the protocol.  The receiver may have other
concerns than absolute rapidity of throughput; for example, power
savings, bandwidth conservation, local processing time, etc. etc.
This, in my mind, simply falls into the realm of dictating
implementation details.  The point is that a receiver delaying that
out-of-order ACK will still perform correctly with respect to protocol
semantics, it may just take a little bit longer.

I am inclined to not change this in 2581bis.  I don't see any major
benefit in doing so, and it does constrain implementors' choices,
while not fixing any serious misinteraction.  Anyone violating that
SHOULD already SHOULD [heh] have a very good reason to be doing so.

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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From tcpm-bounces@ietf.org Sun Feb 25 08:03:43 2007
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Date: Sun, 25 Feb 2007 14:01:08 +0100
From: Michael Welzl <Michael.Welzl@uibk.ac.at>
To: tcpm@ietf.org
Subject: Re: [tcpm] AW: RFC2581bis ooo acking WAS: New I-D posted :
	draft-moncaster-tcpm-rcv-cheat-00
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Folks,

You convinced me. Thanks for the discussion!

Cheers,
Michael


Zitat von Ethan Blanton <eblanton@cs.ohiou.edu>:

> [The "something" below being delaying a dup ACK for a segment received
>  out-of-order using delayed ACKs.]
>
> Michael Welzl spake unto us the following wisdom:
> > Does it make sense to allow a receiver to do something which
> > has no obvious benefit, but clearly leads to undesirable
> > behavior?
>
> Yes, if it does not break the protocol.  The receiver may have other
> concerns than absolute rapidity of throughput; for example, power
> savings, bandwidth conservation, local processing time, etc. etc.
> This, in my mind, simply falls into the realm of dictating
> implementation details.  The point is that a receiver delaying that
> out-of-order ACK will still perform correctly with respect to protocol
> semantics, it may just take a little bit longer.
>
> I am inclined to not change this in 2581bis.  I don't see any major
> benefit in doing so, and it does constrain implementors' choices,
> while not fixing any serious misinteraction.  Anyone violating that
> SHOULD already SHOULD [heh] have a very good reason to be doing so.
>
> Ethan
>
> --
> 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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Subject: RE: [tcpm] AW: RFC2581bis ooo acking WAS: New I-D posted
	:draft-moncaster-tcpm-rcv-cheat-00
Date: Sun, 25 Feb 2007 10:04:39 -0800
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Thread-Topic: [tcpm] AW: RFC2581bis ooo acking WAS: New I-D posted
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Thread-Index: AcdY3Xp0vzFmNpcPS0WyFOClm/8PWQAJAdQw
From: "Anantha Ramaiah \(ananth\)" <ananth@cisco.com>
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Actually my thinking is a bit different here. If there is choice to
pick, and I was given that choice, I would go with MUST ;-) A few
comments inline ( I may be repeating myself here).....=20

<snip>

>=20
> Zitat von Ethan Blanton <eblanton@cs.ohiou.edu>:
>=20
> > [The "something" below being delaying a dup ACK for a=20
> segment received =20
> > out-of-order using delayed ACKs.]
> >
> > Michael Welzl spake unto us the following wisdom:
> > > Does it make sense to allow a receiver to do something=20
> which has no=20
> > > obvious benefit, but clearly leads to undesirable behavior?
> >
> > Yes, if it does not break the protocol.  The receiver may=20
> have other=20
> > concerns than absolute rapidity of throughput; for example, power=20
> > savings, bandwidth conservation, local processing time, etc. etc.
> > This, in my mind, simply falls into the realm of dictating=20
> > implementation details.  The point is that a receiver delaying that=20
> > out-of-order ACK will still perform correctly with respect=20
> to protocol=20
> > semantics, it may just take a little bit longer.

- The question I would ask is: "Is fast retransmission considered a
MUST? Yes, according to RFC 2581. If so I would think the surrounding
mechanisms, esp the ones which has direct influence, which facilitate
the fast-retransmision, can also be considered for a MUST candidate,
like in this case the generation of Duplicate ACK's.=20

- The above example on "power savings, b/w conservation etc.," can be
applied to, in theory, to any packet, not just the duplicate ACK's. I
may sound argumentative here, but my experience ( not theoretical,
coming from working on internet devices/stacks which use TCP) treats the
dupacks as a MUST. Also to my knowledge, I haven't heard of stacks which
don't send or delay ( the latter may be ok, like you point out above)
these duplicate ACKs and would be interested to know if any one of the
widely used stacks permit this either naturally or by means of knob. But
this is my opinion, OMMV.

Thanks,
-Anantha

<snip>

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Subject: Re: [tcpm] AW: RFC2581bis ooo acking WAS: New I-D posted
	:draft-moncaster-tcpm-rcv-cheat-00
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Anantha Ramaiah (ananth) spake unto us the following wisdom:
> > > Yes, if it does not break the protocol.  The receiver may have other=
=20
> > > concerns than absolute rapidity of throughput; for example, power=20
> > > savings, bandwidth conservation, local processing time, etc. etc.
> > > This, in my mind, simply falls into the realm of dictating=20
> > > implementation details.  The point is that a receiver delaying that=
=20
> > > out-of-order ACK will still perform correctly with respect to protocol
> > > semantics, it may just take a little bit longer.
>=20
> - The question I would ask is: "Is fast retransmission considered a
> MUST? Yes, according to RFC 2581. If so I would think the surrounding
> mechanisms, esp the ones which has direct influence, which facilitate
> the fast-retransmision, can also be considered for a MUST candidate,
> like in this case the generation of Duplicate ACK's.=20

I agree; however, the duplicate ACK *will* be generated; it just may
be delayed by as much as 500ms.  (Note that 1122 says that the "delay
MUST be less than 0.5 seconds".)  As several people, including myself,
have said in this thread, it is not at all an issue of *correctness*.
If we were talking about correctness, I would completely agree that
the correct solution is the only option.

Ethan

--=20
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Subject: RE: [tcpm] I-D ACTION:draft-ietf-tcpm-tcpsecure-07.txt 
Date: Sun, 25 Feb 2007 11:35:03 -0800
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From: "Anantha Ramaiah \(ananth\)" <ananth@cisco.com>
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Greetings,
   The main changes from -06- to -07- can be summarised as follows :

- Ted had a an important technical suggestion i.e, to make the
mitigations proposed in the document from MUST to SHOULD. This is
because to echo Ted's statement : "I'm not really keen on making all
existing TCP implementations non-conformant to deal with an attack that
isn't exactly a threat to all the web."  We think it is a very fair
statement and hence changed these mitigations to SHOULD, and the general
intention is to say something like "TCP's SHOULD do this, for TCP's
which do this MUST......"

- editorial, grammatical, typos, correctness stuff. In particular, Ted
and Alfred HInes (Hoenes) provided comments in this area. There were
comments too from other folks. Thanks a lot for all of them.

As a re-cap, the previous revision -06- took care of most of Joe's
comments ( which was sent on the list), revised security considerations
text (thanks to Ted), spell check, grammar and other minor editorial
changes.

Pl review the -07- version. More Comments welcome :-)

Thanks,
-Anantha

> -----Original Message-----
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> Sent: Friday, February 23, 2007 12:50 PM
> To: i-d-announce@ietf.org
> Cc: tcpm@ietf.org
> Subject: [tcpm] I-D ACTION:draft-ietf-tcpm-tcpsecure-07.txt=20
>=20
> A New Internet-Draft is available from the on-line=20
> Internet-Drafts directories.
> This draft is a work item of the TCP Maintenance and Minor=20
> Extensions Working Group of the IETF.
>=20
> 	Title		: Improving TCP's Robustness to Blind=20
> In-Window Attacks
> 	Author(s)	: A. Ramaiah, et al.
> 	Filename	: draft-ietf-tcpm-tcpsecure-07.txt
> 	Pages		: 26
> 	Date		: 2007-2-23
> =09
> TCP has historically been considered protected against spoofed packet
>    injection attacks by relying on the fact that it is difficult to
>    guess the 4-tuple (the source and destination IP addresses and the
>    source and destination ports) in combination with the 32=20
> bit sequence
>    number(s).  A combination of increasing window sizes and=20
> applications
>    using a longer term connections (e.g.  H-323 or Border Gateway
>    Protocol [RFC4271]) have left modern TCP implementation more
>    vulnerable to these types of spoofed packet injection attacks.
>=20
>    Many of these long term TCP applications tend to have=20
> predictable IP
>    addresses and ports which makes it far easier for the 4-tuple to be
>    guessed.  Having guessed the 4-tuple correctly, an attacker can
>    inject a RST, SYN or DATA segment into a TCP connection by=20
> carefully
>    crafting the sequence number of the spoofed segment to be in the
>    current receive window.  This can cause the connection to either
>    abort or possibly cause data corruption.  This document specifies
>    small modifications to the way TCP handles inbound=20
> segments that can
>    reduce the chances of a successful attack.
>=20
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-tcpm-tcpsecure-07.txt
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From tcpm-bounces@ietf.org Mon Feb 26 14:43:38 2007
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	(asymmetry)
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John Heffner wrote:

> Anantha Ramaiah (ananth) wrote:
>
>>> is at odds with RFC 2581, which says:
>>>
>>>> Out-of-order data segments SHOULD be acknowledged immediately, in 
>>>> order to accelerate loss recovery.
>>>
>>> ...although I wonder why this is only a SHOULD, and not a MUST, and 
>>> herewith suggest to change this in 2581bis, unless there's a good 
>>> reason not to do so.
>>
<Snip>

>
> Paths with severe asymmetry may want to thin ACKs.  There were some 
> papers on this in the mid to late 90's coming out of work with 
> satellites and ADSL.
>
These arguments are in RFC 3449, for reference if you wish to go there.

If you expect receivers to ALWAYS generate specific ACKs, the RFC3449 
mitigations will give you some unexpected shocks (i.e. if you try to 
read more into the ACK stream than the TCP sender needed to clock out 
the data).

<Snip>

Gorry Fairhurst

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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		: Defending TCP Against Spoofing Attacks
	Author(s)	: J. Touch
	Filename	: draft-ietf-tcpm-tcp-antispoof-06.txt
	Pages		: 29
	Date		: 2007-2-26
	
Recent analysis of potential attacks on core Internet infrastructure 
   indicates an increased vulnerability of TCP connections to spurious 
   resets (RSTs), sent with forged IP source addresses (spoofing).  TCP 
   has always been susceptible to such RST spoofing attacks, which were 
   indirectly protected by checking that the RST sequence number was 
   inside the current receive window, as well as via the obfuscation of    TCP endpoint and port numbers.  For pairs of well-known endpoints 
   often over predictable port pairs, such as BGP or between web servers 
   and well-known large-scale caches, increases in the path bandwidth-
   delay product of a connection have sufficiently increased the receive 
   window space that off-path third parties can brute-force generate a 
   viable RST sequence number.  The susceptibility to attack increases 
   with the square of the bandwidth, thus presents a significant 
   vulnerability for recent high-speed networks.  This document 
   addresses this vulnerability, discussing proposed solutions at the 
   transport level and their inherent challenges, as well as existing 
   network level solutions and the feasibility of their deployment.  
   This document focuses on vulnerabilities due to spoofed TCP segments, 
   and includes a discussion of related ICMP spoofing attacks on TCP 
   connections.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-tcpm-tcp-antispoof-06.txt

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