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To: "Anantha Ramaiah (ananth)" <ananth@cisco.com>
From: Mark Allman <mallman@icir.org>
Subject: Re: [tcpm] tcpsecure: how strong to recommend? 
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Hyperbole elided, but still trying to understand ...

> but
> saying that "SHOULD doesn't give leeway not to do something", is
> something which I would want to dis-agree.

If tcpsecure is a SHOULD then what would be an invalid reason for me to
exclude it from my implementation?

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From tcpm-bounces@ietf.org Mon Oct 01 10:19:50 2007
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From: Pasi Sarolahti <pasi.sarolahti@nokia.com>
Subject: Re: [tcpm] Re: draft-ietf-tcpm-rfc4138bis-00
Date: Mon, 1 Oct 2007 17:15:42 +0300
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On Sep 26, 2007, at 16:52, ext Ilpo J=E4rvinen wrote:

>> Also, the SACK version of F-RTO has not been included in all
>> implementations that have the basic version
>> (for example Linux distribution has had only the basic version).
>
> ...That's only partially valid because since 2.6.22 Linux has included
> both versions (though both off by default). Its SACK version includes
> features discussed in 4138's Appendix B. =46rom 2.6.24 onward it will
> enable SACK version by default.

So I had outdated information. Thanks for keeping us updated!

Ok, we were too hasty in removing the SACK variant from the draft. =20
The feedback so far, on and off the mailing list, seem to be in favor =20=

of keeping it. We'll add it back for the next version.

> Quite clearly using SACK blocks improves FRTO's accuracy in =20
> detection as
> they are rather strong indication what really arrived. However, it =20
> adds
> some challenges to aggressive response usage because then lying in =20
> SACK
> blocks should also be addressed in Security Considerations, =20
> especially as
> "because the sender will have an incorrect state, it cannot retransmit
> the segment that has never reached the receiver" does not hold like it
> does with SND.UNA. Yet, as long as FRTO is accompanied with a =20
> conservative
> response, I don't see how the receiver could have meaningful =20
> benefits from
> lying, whereas unnecessary retransmissions are more accurately =20
> mitigated.

This reminds me of another issue mentioned in the Chicago meeting =20
that might deserve discussion: what should we say about the response =20
in rfc4138bis? The current draft refers to the Eifel Response and to =20
draft-allman-rto-backoff as possible response algorithms, but also =20
says that if neither of these is implemented, the TCP sender should =20
respond conservatively to spurious timeout, applying the TCP =20
congestion control specification. I wonder if people prefer more =20
definite words here about the response, or is it just enough to refer =20=

to the known choices, as we now have done. What you say above seems =20
to indicate that at least with SACK we might want to prefer a =20
conservative response.

- Pasi



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Subject: RE: [tcpm] tcpsecure: how strong to recommend? 
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=20
>=20
> Hyperbole elided, but still trying to understand ...
>=20
> > but
> > saying that "SHOULD doesn't give leeway not to do something", is=20
> > something which I would want to dis-agree.
>=20
> If tcpsecure is a SHOULD then what would be an invalid reason=20
> for me to exclude it from my implementation?

It is difficult to say since the implementer is given the choice whether
or not to include/exclude it. Yes, both SHOULD and MAY provides that
luxury. The only difference is the strength of the advise offered by
both these adjectives which needs to be viewed from a recommendation
perspective. If you think that these are generally good to have then use
SHOULD else these are really optional ones use MAY. The bottomline is
that unless it is tagged as MUST, I believe we have already given the
flexibility, the key is how "strong" we would want to recommend and that
is what we are trying to do and get the consensus. Atleast this *my
understanding*. I expressed my opinion and other have expressed theirs.=20

Also, I am trying to understand something here since I wasn't sure how
situations like this have been dealt in the past. Actually, I am trying
to understand the algorithm used to arrive consensus here?. I was
assuming that after the "hums were sought" and the email was sent both
offering 3 choices, now after getting the responses the next step would
be to derive something out of that once we have reached enough responses
or a stipulated time expires if one exists. If we believe there isn't
even a rough consensus after the responses so far, what is the next
step? Curious.

-Anantha=20


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From tcpm-bounces@ietf.org Mon Oct 01 12:35:43 2007
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To: "Anantha Ramaiah (ananth)" <ananth@cisco.com>
From: Mark Allman <mallman@icir.org>
Subject: Re: [tcpm] tcpsecure: how strong to recommend? 
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> If we believe there isn't
> even a rough consensus after the responses so far, what is the next
> step? Curious.

Per my previous email:

  It seems to me that this discussion is really divergent because there
  is no applicability statement in the document, per Lars' comment.  I
  wonder if you guys could go off and generate such a statement and then
  we could re-visit this question.  I think that would factor things
  into a question of "where" this is applicable and then how strongly we
  want to advocate these mitigations within that context.  Is that
  reasonable?

(From which my understanding was that you guys were going to generate
said statement.)

allman




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I normally just lurk here silently, and I certainly wasn't at Chicago or 
any of the other meetings discussing this.  But it strikes me that people 
are getting caught up in language problems and losing sight of what is 
desired.  In particular there is confusion between discussing whether 
implementing tcpsecure itself is mandatory / recommended / optional vs. 
whether some feature of tcpsecure is mandatory / recommended / optional, 
given that tcpsecure itself is implemented.

Clearly there will be implementations that do not implement 
tcpsecure.  Historical implementations.  Those that achieve similar ends 
through different means (e.g. IPSEC, MD5) and therefore have no need or use 
for tcpsecure.

Clearly there are implementations that need or want tcpsecure, BGP being 
the prominent example.

So there are two implementation variations that we consider necessary.  Is 
there any need or point in more?  Remember that compatibility problems, 
testing, etc. scale as (at least) NxN.  There is enormous practical 
incentive to minimize the number of variations that are implemented and 
deployed.

I contend that the specifications or RFCs should allow exactly two 
variations.  Those implementations that ignore and do not implement 
tcpsecure at all.  And those that implement all of it.

I think the way to accomplish that is to include an applicability statement 
saying that implementing tcpsecure is optional.  An implementation MAY or 
MAY NOT implement it.  And giving guidance on when one should or should 
not.  Probably something along the lines of saying one should implement it 
unless the same ends are accomplished by other means, and giving several 
examples of other means.

But if an implementation does choose to implement tcpsecure, then it MUST 
implement all of it -- the features, protocol behavior, etc. are described 
in the RFC using MUST.  Conversely, if an implementation chooses not to 
implement tcpsecure, then it MUST NOT implement any part of it.  Either 
implement all of tcpsecure or none of it.  Yes, this requires careful 
phrasing, but it can be done.


I am assuming that all the features included in tcpsecure are considered 
necessary or desirable for BGP.  If any are not, shouldn't they be excluded 
on the grounds that no one is ever likely to implement them?

One might contemplate the impact of the IETF requirement for independently 
developed, interoperable implementations.  The three tcpsecure (three 
choices of MAY vs. SHOULD that are being discussed) interact to some 
degree.  If each are described using MAY or SHOULD, we get eight 
implementation variations.  Each needing two independent implementations 
means at least sixteen implementation, which is absurd.  Does anyone really 
want to allow some arguably ill-informed implementors to pick and choose, 
as opposed to saying all or none?  Isn't it our responsibility as 
architects (who claim to know what we are doing :-) to minimize variations, 
not unnecessarily expand them?

Caveat: my primary experience is with the T10 and T11 standards (SCSI and 
friends), I apologize for my errors with IETF terminology and practice.

Dropping back into my hole to resume lurking.


Edward A. Gardner               eag at ophidian dot com
Ophidian Designs                719 593-8866
1262 Hofstead Terrace
Colorado Springs, CO  80907



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Subject: RE: [tcpm] tcpsecure: how strong to recommend? 
Date: Mon, 1 Oct 2007 14:55:24 -0700
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Thread-Topic: [tcpm] tcpsecure: how strong to recommend? 
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From: "Mitesh Dalal (mdalal)" <mdalal@cisco.com>
To: "Edward A. Gardner" <eag@ophidian.com>, <tcpm@ietf.org>
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well said Edward! I concur with your take on this and was alluding to
the same ideas that you mentioned below. To highlight your comment
again:

> In particular=20
> there is confusion between discussing whether implementing=20
> tcpsecure itself is mandatory / recommended / optional vs.=20
> whether some feature of tcpsecure is mandatory / recommended=20
> / optional, given that tcpsecure itself is implemented.


I think it does captures the debate we are having here. My
opinion is that if we view these changes as Updates to RFC793, then
they may be perceived as mandatory changes and again open discussions
around IPR, MAY/SHOULD/MUST language and so on. However if we have
a  banner statement stating that implementing these changes are
recommended/optional and enumerate situations in which they are=20
helpful this should alleviate a lot of opposing view.  And
once we have this statement and if a tcp implementation does decide
to have this feature then it MUST be this way for the feature to work
correctly.

So to reiterate to the group, we need an applicability statement that
highlights the implementations/deployments/applications that are
RECOMMENDED to have this fix, with others being OPTIONAL. However,
the features/changes themselves being MUST (or SHOULD, if that's what
it would take for a consensus). Is the group in agreement here?

my 0.02
Mitesh

 =20

> -----Original Message-----
> From: Edward A. Gardner [mailto:eag@ophidian.com]=20
> Sent: Monday, October 01, 2007 11:16 AM
> To: tcpm@ietf.org
> Subject: Re: [tcpm] tcpsecure: how strong to recommend?=20
>=20
> I normally just lurk here silently, and I certainly wasn't at=20
> Chicago or any of the other meetings discussing this.  But it=20
> strikes me that people are getting caught up in language=20
> problems and losing sight of what is desired.  In particular=20
> there is confusion between discussing whether implementing=20
> tcpsecure itself is mandatory / recommended / optional vs.=20
> whether some feature of tcpsecure is mandatory / recommended=20
> / optional, given that tcpsecure itself is implemented.
>=20
> Clearly there will be implementations that do not implement=20
> tcpsecure.  Historical implementations.  Those that achieve=20
> similar ends through different means (e.g. IPSEC, MD5) and=20
> therefore have no need or use for tcpsecure.
>=20
> Clearly there are implementations that need or want=20
> tcpsecure, BGP being the prominent example.
>=20
> So there are two implementation variations that we consider=20
> necessary.  Is there any need or point in more?  Remember=20
> that compatibility problems, testing, etc. scale as (at=20
> least) NxN.  There is enormous practical incentive to=20
> minimize the number of variations that are implemented and deployed.
>=20
> I contend that the specifications or RFCs should allow=20
> exactly two variations.  Those implementations that ignore=20
> and do not implement tcpsecure at all.  And those that=20
> implement all of it.
>=20
> I think the way to accomplish that is to include an=20
> applicability statement saying that implementing tcpsecure is=20
> optional.  An implementation MAY or MAY NOT implement it. =20
> And giving guidance on when one should or should not. =20
> Probably something along the lines of saying one should=20
> implement it unless the same ends are accomplished by other=20
> means, and giving several examples of other means.
>=20
> But if an implementation does choose to implement tcpsecure,=20
> then it MUST implement all of it -- the features, protocol=20
> behavior, etc. are described in the RFC using MUST. =20
> Conversely, if an implementation chooses not to implement=20
> tcpsecure, then it MUST NOT implement any part of it.  Either=20
> implement all of tcpsecure or none of it.  Yes, this requires=20
> careful phrasing, but it can be done.
>=20
>=20
> I am assuming that all the features included in tcpsecure are=20
> considered necessary or desirable for BGP.  If any are not,=20
> shouldn't they be excluded on the grounds that no one is ever=20
> likely to implement them?
>=20
> One might contemplate the impact of the IETF requirement for=20
> independently developed, interoperable implementations.  The=20
> three tcpsecure (three choices of MAY vs. SHOULD that are=20
> being discussed) interact to some degree.  If each are=20
> described using MAY or SHOULD, we get eight implementation=20
> variations.  Each needing two independent implementations=20
> means at least sixteen implementation, which is absurd.  Does=20
> anyone really want to allow some arguably ill-informed=20
> implementors to pick and choose, as opposed to saying all or=20
> none?  Isn't it our responsibility as architects (who claim=20
> to know what we are doing :-) to minimize variations, not=20
> unnecessarily expand them?
>=20
> Caveat: my primary experience is with the T10 and T11=20
> standards (SCSI and friends), I apologize for my errors with=20
> IETF terminology and practice.
>=20
> Dropping back into my hole to resume lurking.
>=20
>=20
> Edward A. Gardner               eag at ophidian dot com
> Ophidian Designs                719 593-8866
> 1262 Hofstead Terrace
> Colorado Springs, CO  80907
>=20
>=20
>=20
> _______________________________________________
> tcpm mailing list
> tcpm@ietf.org
> https://www1.ietf.org/mailman/listinfo/tcpm
>=20


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From tcpm-bounces@ietf.org Mon Oct 01 19:16:22 2007
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Mitesh Dalal (mdalal) wrote:
> well said Edward! I concur with your take on this and was alluding to
> the same ideas that you mentioned below. To highlight your comment
> again:
>=20
>> In particular=20
>> there is confusion between discussing whether implementing=20
>> tcpsecure itself is mandatory / recommended / optional vs.=20
>> whether some feature of tcpsecure is mandatory / recommended=20
>> / optional, given that tcpsecure itself is implemented.
>=20
>=20
> I think it does captures the debate we are having here. My
> opinion is that if we view these changes as Updates to RFC793, then
> they may be perceived as mandatory changes and again open discussions
> around IPR, MAY/SHOULD/MUST language and so on. However if we have
> a  banner statement stating that implementing these changes are
> recommended/optional and enumerate situations in which they are=20
> helpful this should alleviate a lot of opposing view.  And
> once we have this statement and if a tcp implementation does decide
> to have this feature then it MUST be this way for the feature to work
> correctly.\

I think we agree that there are two levels of language needed:

1) for the entire system
	for which I think MAY is appropriate,
	and the conditions when useful would be beneficial to note

2) for the components, once you decide to implement
	MUST/MAY/SHOULDs as needed there. there seem like there are
	a few MAYs for various parts, MUSTs for others;
	it's not all MUSTs, but I don't think there will be
	deep discussions about this

Joe



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Subject: Re: [tcpm] tcpsecure: how strong to recommend? 
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On 2007-10-1, at 19:23, ext Anantha Ramaiah (ananth) wrote:
>> If tcpsecure is a SHOULD then what would be an invalid reason
>> for me to exclude it from my implementation?
>
> It is difficult to say since the implementer is given the choice  
> whether
> or not to include/exclude it.

RFC2026 says:

3. SHOULD   This word, or the adjective "RECOMMENDED", mean that there
    may exist valid reasons in particular circumstances to ignore a
    particular item, but the full implications must be understood and
    carefully weighed before choosing a different course.

Some in the IESG have argued that "(...) implications must be  
understood and carefully weighed" means that drafts needs to  
explicitly discuss under what conditions it is OK to disregard a  
SHOULD ("choosing a different course.")

Obviously, few drafts do this. But because we're hung up on what to  
do with this particular draft, following RFC2026 more closely might  
help us make progress.

Lars
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Subject: RE: [tcpm] tcpsecure: how strong to recommend? 
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Thread-Topic: [tcpm] tcpsecure: how strong to recommend? 
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From: "Anantha Ramaiah (ananth)" <ananth@cisco.com>
To: "Mitesh Dalal (mdalal)" <mdalal@cisco.com>,
	"Edward A. Gardner" <eag@ophidian.com>, <tcpm@ietf.org>
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=20
> So to reiterate to the group, we need an applicability=20
> statement that highlights the=20
> implementations/deployments/applications that are RECOMMENDED=20
> to have this fix, with others being OPTIONAL. However, the=20
> features/changes themselves being MUST (or SHOULD, if that's=20
> what it would take for a consensus). Is the group in agreement here?

++;=20
(with the author hat off :-)

-Anantha


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Anantha Ramaiah (ananth) wrote:
> =20
>> So to reiterate to the group, we need an applicability=20
>> statement that highlights the=20
>> implementations/deployments/applications that are RECOMMENDED=20
>> to have this fix, with others being OPTIONAL. However, the=20
>> features/changes themselves being MUST (or SHOULD, if that's=20
>> what it would take for a consensus). Is the group in agreement here?
>=20
> ++;=20
> (with the author hat off :-)

As a way forward, the following captures the SHOULD/SHOULD/MAY which was
 most supported in Chicago:

-------------------------------------------------------------------------=
--

tcpsecure SHOULD be implemented in TCP stacks supporting router.
Notable exceptions include deployments where routers are known to use
other antispoofing protection, e.g., IPsec, TCP/MD5 and its successors.

tcpsecure MAY be implemented in other TCP stacks.

----
within tcpsecure:

RST protection MUST be supported

SYN protection MUST be supported

data segment (i.e., non-RST, non-SYN) protection MAY be supported

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







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> As a way forward, the following captures the=20
> SHOULD/SHOULD/MAY which was
>  most supported in Chicago:
>=20
> --------------------------------------------------------------
> -------------
>=20
> tcpsecure SHOULD be implemented in TCP stacks supporting router.
> Notable exceptions include deployments where routers are known to use
> other antispoofing protection, e.g., IPsec, TCP/MD5 and its=20
> successors.
>=20
> tcpsecure MAY be implemented in other TCP stacks.
>=20
> ----


> within tcpsecure:
>=20
> RST protection MUST be supported
>=20
> SYN protection MUST be supported
>=20
> data segment (i.e., non-RST, non-SYN) protection MAY be supported
>=20
> -------------------------------------------

I think you got it wrong.=20

Most supported in Chicago was about [SHOULD/SHOULD/MAY] WITHOUT the
applicability statement in place. It was "STANDALONE" question raised by
the chairs for which the consensus seemed to be in favour of S/S/M.  Now
WITH the applicability statement in place, it completely changes the
entire equation. To quote Mark's email :

BEGIN QUOTE
> It seems to me that this discussion is really divergent because there
> is no applicability statement in the document, per Lars' comment.  I
> wonder if you guys could go off and generate such a statement and=20
> then we could re-visit this question.  I think that would factor
things
> into a question of "where" this is applicable and then how strongly we
> want to advocate these mitigations within that context.  Is that
> reasonable?
END QUOTE

So, the game plan moving forward is, to generate the AS and revisit the
mitigation strengths. Also, just in case you missed, some of responses
in this list have indicated no issues with "MUST'ing" all the
mitigations provided there is a proper applicability statement in place.

-Anantha


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From tcpm-bounces@ietf.org Tue Oct 02 16:32:01 2007
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Notes below....

Anantha Ramaiah (ananth) wrote:
>> As a way forward, the following captures the=20
>> SHOULD/SHOULD/MAY which was
>>  most supported in Chicago:
>>
>> --------------------------------------------------------------
>> -------------
>>
>> tcpsecure SHOULD be implemented in TCP stacks supporting router.
>> Notable exceptions include deployments where routers are known to use
>> other antispoofing protection, e.g., IPsec, TCP/MD5 and its=20
>> successors.
>>
>> tcpsecure MAY be implemented in other TCP stacks.
>>
>> ----
>=20
>=20
>> within tcpsecure:
>>
>> RST protection MUST be supported
>>
>> SYN protection MUST be supported
>>
>> data segment (i.e., non-RST, non-SYN) protection MAY be supported
>>
>> -------------------------------------------
>=20
> I think you got it wrong.=20
>=20
> Most supported in Chicago was about [SHOULD/SHOULD/MAY] WITHOUT the
> applicability statement in place.

Agreed. I was trying to capture Chicago in the bottom portion.

> It was "STANDALONE" question raised by
> the chairs for which the consensus seemed to be in favour of S/S/M.  No=
w
> WITH the applicability statement in place, it completely changes the
> entire equation.=20

I don't think it does. Even with a qualified SHOULD on the overall set,
data segment protection is still a MAY. What it changes is the SHOULDs
on the RST and SYN to MUSTs, to say that you shouldn't implement one or
the other as desired *if* you decide to implement the set.

=2E..
> So, the game plan moving forward is, to generate the AS and revisit the=

> mitigation strengths.=20

I tried to do that above.

> Also, just in case you missed, some of responses
> in this list have indicated no issues with "MUST'ing" all the
> mitigations provided there is a proper applicability statement in place=
=2E

I know - that was one of the set discussed. OK, so here's what I'm
specifically asking:

1) is the AS above reasonable?

2) regarding the RST/SYN/data components:
	- are the MUST/MUST/MAY above reasonable?
	- Or should they be SHOULD/SHOULD/MAY?
	- or something else

Joe


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Subject: RE: [tcpm] tcpsecure: how strong to recommend?
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Thread-Topic: [tcpm] tcpsecure: how strong to recommend?
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Pl see inline comments...
=20
> >> tcpsecure SHOULD be implemented in TCP stacks supporting router.
> >> Notable exceptions include deployments where routers are=20
> known to use=20
> >> other antispoofing protection, e.g., IPsec, TCP/MD5 and its=20
> >> successors.
> >>
> >> tcpsecure MAY be implemented in other TCP stacks.
> >>

<snip>

>=20
> > It was "STANDALONE" question raised by the chairs for which the=20
> > consensus seemed to be in favour of S/S/M.  Now WITH the=20
> applicability=20
> > statement in place, it completely changes the entire equation.
>=20
> I don't think it does. Even with a qualified SHOULD on the=20
> overall set, data segment protection is still a MAY. What it=20
> changes is the SHOULDs on the RST and SYN to MUSTs, to say=20
> that you shouldn't implement one or the other as desired *if*=20
> you decide to implement the set.

I dis-agree. All the reasoning so far with the "chosen" adjectives for
the individual mitigations are without the applicability statement in
place.
<snip>

>=20
> > Also, just in case you missed, some of responses in this list have=20
> > indicated no issues with "MUST'ing" all the mitigations=20
> provided there=20
> > is a proper applicability statement in place.
>=20
> I know - that was one of the set discussed. OK, so here's=20
> what I'm specifically asking:
>=20
> 1) is the AS above reasonable?

Well, I (some others also pointed out) wouldn't mention router, since
these mitigations can be/are used in devices not routers per se. But we
can use this text as a reference for generating the AS. Others may want
to comment here as well.

>=20
> 2) regarding the RST/SYN/data components:
> 	- are the MUST/MUST/MAY above reasonable?
> 	- Or should they be SHOULD/SHOULD/MAY?
> 	- or something else

Something else. To me, all the mitigations are equally important and
hence they  classified as MUST after the AS is in place.

-Anantha
>=20
> Joe
>=20
>=20


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Anantha Ramaiah (ananth) wrote:
=2E..
>> 1) is the AS above reasonable?
>=20
> Well, I (some others also pointed out) wouldn't mention router, since
> these mitigations can be/are used in devices not routers per se. But we=

> can use this text as a reference for generating the AS. Others may want=

> to comment here as well.

Can you suggest a term that's useful? How about something specific:

"Devices with services where both endpoints are typically known, e.g.,
BGP."

>> 2) regarding the RST/SYN/data components:
>> 	- are the MUST/MUST/MAY above reasonable?
>> 	- Or should they be SHOULD/SHOULD/MAY?
>> 	- or something else
>=20
> Something else. To me, all the mitigations are equally important and
> hence they  classified as MUST after the AS is in place.

OK, so here are some options:

	a) MUST/MUST/MUST
	b) MUST/MUST/SHOULD
	c) MUST/MUST/MAY

I don't quite understand why we would go with SHOULD for either RST or
SYN individually; if anyone can, it'd be useful to make the case. We can
mark Anantha as being in favor of (a). I would interpret the WG as
*probably* being in favor of (c), since if data was a MAY unqualified,
then at best it's a MAY qualified, BUT I'm not trying to make that case
myself.

I prefer (c), but that's just my individual perspective. I wouldn't care
much which of a-c is used.

Joe


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Subject: [tcpm] Re: [Tsvwg] consensus call:
	draft-larsen-tsvwg-port-randomization to WG item
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Hi,

my reading of the list discussion is that we do have consensus for  
TSVWG to take this on this work item with draft-larsen-tsvwg-port- 
randomization as the basis, i.e., option (1) below.

Authors: Please submit the next revision as draft-ietf-tsvwg-port- 
randomization-00 with a target of Proposed Standard. You should also  
review the comments that have been made on the list and incorporate  
them into the revision. For example, several people suggested that  
the document should more prominently recommend one randomization  
scheme (or a small number of alternatives). The analysis of the pros  
and cons of currently implemented schemes that is the bulk of the  
current document should maybe become an informative appendix.

DCCP & TCPM working groups: The continued discussion around this  
draft will occur on the TSVWG list only, to minimize cross-posting.  
There will be a joint WG last call eventually, but please participate  
in the ongoing discussion on TSVWG until then.

On 2007-9-12, at 11:38, ext Lars Eggert wrote:
> Please comment on the tsvwg@ietf.org list (reply-to set  
> accordingly), even if you have already commented be on this draft  
> in the past.
>
> To make it easy for your chairs, please start your email by  
> choosing ONE of the options below, followed by any detailed  
> comments you'd like to make:
>
>   (1) TSVWG should take on this work item, with this document as  
> the basis
>   (2) TSVWG should take on this work item, but NOT with this  
> document as the basis
>   (3) TSVWG should NOT take on this work item
>   (4) don't care

Lars
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From tcpm-bounces@ietf.org Wed Oct 03 13:27:40 2007
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From: Ted Faber <faber@ISI.EDU>
To: Joe Touch <touch@ISI.EDU>
Subject: Re: [tcpm] tcpsecure: how strong to recommend?
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On Sat, Sep 29, 2007 at 11:19:38PM -0700, Joe Touch wrote:
> Anantha Ramaiah (ananth) wrote:
> > I will have to dis-agree to this since my viewpoint is different. TCP
> > secure adds robustness to the processing of certain TCP segments, which
> > in turn helps to counter *some* spoofing attacks. Calling it as an
> > authentication scheme seems too far-fetched.
>=20
> You are making an assertion about whether you believe the packet is
> spoofed or not based on its content matching what you expect from the
> true endpoint.
>=20
> That is called authentication. Weak, but still authentication.

Acting without a chair hat, I disagree.  The packet is being categorized
as suspicious, for example, it could have been spoofed, corrupted,
significantly delayed, whatever.  I see the ACK is an attempt to
synchronize the endpoints' states, not an attempt to autenticate the
peer.  The question being asked is closer to "what's going on on your
end?" than "who sent this packet?"

While some of the discussion has definitely leaned toward security and
authentication terminology, I don't think that's the essence of the
proposed system.

--=20
Ted Faber
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Ted Faber wrote:
> On Sat, Sep 29, 2007 at 11:19:38PM -0700, Joe Touch wrote:
>> Anantha Ramaiah (ananth) wrote:
>>> I will have to dis-agree to this since my viewpoint is different. TCP=

>>> secure adds robustness to the processing of certain TCP segments, whi=
ch
>>> in turn helps to counter *some* spoofing attacks. Calling it as an
>>> authentication scheme seems too far-fetched.
>> You are making an assertion about whether you believe the packet is
>> spoofed or not based on its content matching what you expect from the
>> true endpoint.
>>
>> That is called authentication. Weak, but still authentication.
>=20
> Acting without a chair hat, I disagree.  The packet is being categorize=
d
> as suspicious, for example, it could have been spoofed, corrupted,
> significantly delayed, whatever.  I see the ACK is an attempt to
> synchronize the endpoints' states, not an attempt to autenticate the
> peer.  The question being asked is closer to "what's going on on your
> end?" than "who sent this packet?"
>=20
> While some of the discussion has definitely leaned toward security and
> authentication terminology, I don't think that's the essence of the
> proposed system.

OK, so that makes it even more strange. You're requiring a unilateral
reset (RST) to be issued only when the endpoint states are precisely
aligned (no outstanding unack'd segments).

Why?

What does that tell you about the other end that you didn't know? What
purpose relevant to resetting a connection does synchronizing state
serve if not to authenticate the other end?

If not, then this is an even more bizarre requirement on an otherwise
very simple mechanism.

Joe


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From: Ted Faber <faber@ISI.EDU>
To: Joe Touch <touch@ISI.EDU>
Subject: Re: [tcpm] tcpsecure: how strong to recommend?
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On Wed, Oct 03, 2007 at 10:29:09AM -0700, Joe Touch wrote:
>=20
>=20
> Ted Faber wrote:
> > Acting without a chair hat, I disagree.  The packet is being categorized
> > as suspicious, for example, it could have been spoofed, corrupted,
> > significantly delayed, whatever.  I see the ACK is an attempt to
> > synchronize the endpoints' states, not an attempt to autenticate the
> > peer.  The question being asked is closer to "what's going on on your
> > end?" than "who sent this packet?"
[snip]
> >=20
> OK, so that makes it even more strange. You're requiring a unilateral
> reset (RST) to be issued only when the endpoint states are precisely
> aligned (no outstanding unack'd segments).
>=20
> Why?

To make it unlikely that spoofed, delayed, erroneously generated, or
otherwise invalid packets will terminate an existing connection.

A correct implementation intentionally resetting will quickly become
synchronized even if the first RST was not from a perfectly synched
state.  The reset is potentially slower, but a poorly performing abort
facility doesn't upset me much.

Do you disagree with the characterization of the system or its
operation?

>=20
> What does that tell you about the other end that you didn't know? What
> purpose relevant to resetting a connection does synchronizing state
> serve if not to authenticate the other end?

If the RST that triggered the ACK was from another connection, the
TCPsecure exchange tells you that the other end of this connection has
not reset.  No new information about the identity of that endpoint has
been asserted, simply that the end that got your ACK has state closely
enough synchronized with yours to make a sensical reply.  The identity
of the sender of the RST remains unknown (though if the other end *has*
reset, the sender is likely the other end).

I don't know anything new about the identity of anyone in the
communication; I just don't see authentication there.

> If not, then this is an even more bizarre requirement on an otherwise
> very simple mechanism.

Yeah, this has the feel of us talking past each other.

--=20
Ted Faber
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Subject: Re: [tcpm] tcpsecure: how strong to recommend?
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Ted Faber wrote:
> On Wed, Oct 03, 2007 at 10:29:09AM -0700, Joe Touch wrote:
>>
>> Ted Faber wrote:
>>> Acting without a chair hat, I disagree.  The packet is being categori=
zed
>>> as suspicious, for example, it could have been spoofed, corrupted,
>>> significantly delayed, whatever.  I see the ACK is an attempt to
>>> synchronize the endpoints' states, not an attempt to autenticate the
>>> peer.  The question being asked is closer to "what's going on on your=

>>> end?" than "who sent this packet?"
> [snip]
>> OK, so that makes it even more strange. You're requiring a unilateral
>> reset (RST) to be issued only when the endpoint states are precisely
>> aligned (no outstanding unack'd segments).
>>
>> Why?
>=20
> To make it unlikely that spoofed, delayed, erroneously generated, or
> otherwise invalid packets will terminate an existing connection.
>=20
> A correct implementation intentionally resetting will quickly become
> synchronized even if the first RST was not from a perfectly synched
> state.  The reset is potentially slower, but a poorly performing abort
> facility doesn't upset me much.
>=20
> Do you disagree with the characterization of the system or its
> operation?

Asking to 'synchronize with the other end' is authentication, especially
when the only purpose behind it is to determine whether a spoofed packet
is generated.

It's insincere to rationalize events like erroneously generated or
delayed RSTs into this discussion. Both kinds of RSTs are just as valid
as correctly generated, timely ones, as far as the TCP spec is
concerned, unless you're concerned they're beings spoofed.

>> What does that tell you about the other end that you didn't know? What=

>> purpose relevant to resetting a connection does synchronizing state
>> serve if not to authenticate the other end?
>=20
> If the RST that triggered the ACK was from another connection, the
> TCPsecure exchange tells you that the other end of this connection has
> not reset. =20

That presumes that either the RST is floating in the network more than 2
MSLs or that your end decided to accept a connection it shouldn't have
since it didn't correctly wait 2 MSLs from a previous connection.

If we're going to patch TCP to fix >2MSL errors, there's a lot more work
than this to be done. If your end is broken, then expecting this patch
to correct the situation is not productive.

> No new information about the identity of that endpoint has
> been asserted, simply that the end that got your ACK has state closely
> enough synchronized with yours to make a sensical reply.  The identity
> of the sender of the RST remains unknown (though if the other end *has*=

> reset, the sender is likely the other end).

The identity of the other end is defined by the synchronization process
- it's a TCP endpoint with which your ACKs synchronize state.

> I don't know anything new about the identity of anyone in the
> communication; I just don't see authentication there.

You know it's the other end with which you can synchronize state. That's
identity.

Joe


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From: Ted Faber <faber@ISI.EDU>
To: Joe Touch <touch@ISI.EDU>
Subject: Re: [tcpm] tcpsecure: how strong to recommend?
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On Wed, Oct 03, 2007 at 11:37:39AM -0700, Joe Touch wrote:
> Ted Faber wrote:
> > On Wed, Oct 03, 2007 at 10:29:09AM -0700, Joe Touch wrote:
> >>
> >> Ted Faber wrote:
> >>> Acting without a chair hat, I disagree.  The packet is being categori=
zed
> >>> as suspicious, for example, it could have been spoofed, corrupted,
> >>> significantly delayed, whatever.  I see the ACK is an attempt to
> >>> synchronize the endpoints' states, not an attempt to autenticate the
> >>> peer.  The question being asked is closer to "what's going on on your
> >>> end?" than "who sent this packet?"
> > [snip]
> >> OK, so that makes it even more strange. You're requiring a unilateral
> >> reset (RST) to be issued only when the endpoint states are precisely
> >> aligned (no outstanding unack'd segments).
> >>
> >> Why?
> >=20
> > To make it unlikely that spoofed, delayed, erroneously generated, or
> > otherwise invalid packets will terminate an existing connection.
> >=20
> > A correct implementation intentionally resetting will quickly become
> > synchronized even if the first RST was not from a perfectly synched
> > state.  The reset is potentially slower, but a poorly performing abort
> > facility doesn't upset me much.
> >=20
> > Do you disagree with the characterization of the system or its
> > operation?
>=20
> Asking to 'synchronize with the other end' is authentication, especially
> when the only purpose behind it is to determine whether a spoofed packet
> is generated.

"Did you say that?" is a fundamentally different question that "Who said
that" in my mind.  Asking the first in this context is both more useful
and more practical to determine.

If you want to think of them both as authentication, I'm not going to
try any more to convince you not to.

>=20
> It's insincere to rationalize events like erroneously generated or
> delayed RSTs into this discussion. Both kinds of RSTs are just as valid
> as correctly generated, timely ones, as far as the TCP spec is
> concerned, unless you're concerned they're beings spoofed.

I'm perfectly sincere about it.  I think pointing out that the technique
mitigates several possibilities emphasizes its value from an engineering
perspective.

I can't argue with your perfectly correct statement that a compliant TCP
will treat the spoofed, out of date, erroneously generated, or damaged
packets as correct and reset the connection.  The argument is whether
that behavior is useful or not. =20

I think it's not, but YMMV.

>=20
> >> What does that tell you about the other end that you didn't know? What
> >> purpose relevant to resetting a connection does synchronizing state
> >> serve if not to authenticate the other end?
> >=20
> > If the RST that triggered the ACK was from another connection, the
> > TCPsecure exchange tells you that the other end of this connection has
> > not reset. =20
>=20
> That presumes that either the RST is floating in the network more than 2
> MSLs or that your end decided to accept a connection it shouldn't have
> since it didn't correctly wait 2 MSLs from a previous connection.

Again, technically correct.  I think that addressing the eventuality that
(among other things) MSL is incorrectly estimated or configured with an
extra packet exchange is good engineering.

Yes, I know that an "engineering decision" is often difficult to tell
from an "egregious hack."  I think that this design is the former.

>=20
> If we're going to patch TCP to fix >2MSL errors, there's a lot more work
> than this to be done. If your end is broken, then expecting this patch
> to correct the situation is not productive.

Again, engineering vs. protocol design.

IMHO, we're already addressing unlikely events - MSL violation or
off-path spoofed RST with known addresses and ports.  I think that the
cost of one packet exchange to validate that the connection is not about
to be terminated by an unlikely event is a reasonable engineering
choice.  I do not characterize this as provably correcting a protocol
flaw.

>=20
> > No new information about the identity of that endpoint has
> > been asserted, simply that the end that got your ACK has state closely
> > enough synchronized with yours to make a sensical reply.  The identity
> > of the sender of the RST remains unknown (though if the other end *has*
> > reset, the sender is likely the other end).
>=20
> The identity of the other end is defined by the synchronization process
> - it's a TCP endpoint with which your ACKs synchronize state.
>=20
> > I don't know anything new about the identity of anyone in the
> > communication; I just don't see authentication there.
>=20
> You know it's the other end with which you can synchronize state. That's
> identity.

I think we're arguing over what to name the process.  This would be fun
to do in person, but I don't think it's advancing the discussion of the
system, or in particular of the discussion about the guidance to
implementers we're trying to decide on.

--=20
Ted Faber
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To: Ted Faber <faber@ISI.EDU>
Subject: Re: [tcpm] tcpsecure: how strong to recommend?
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Quoting Ted Faber <faber@ISI.EDU>:

> > Asking to 'synchronize with the other end' is authentication, especially
> > when the only purpose behind it is to determine whether a spoofed packet
> > is generated.
> 
> "Did you say that?" is a fundamentally different question that "Who said
> that" in my mind.  Asking the first in this context is both more useful
> and more practical to determine.
> 
> If you want to think of them both as authentication, I'm not going to
> try any more to convince you not to.

The only difference between "did you say that" and "who said that" is who is
providing the authentication. The latter can be authenticated by a third party,
where the first is done exclusively by the other end of the connection. I hope
that at least helps explain why I see these as equivalent, even if not everyone
agrees with it.

> > It's insincere to rationalize events like erroneously generated or
> > delayed RSTs into this discussion. Both kinds of RSTs are just as valid
> > as correctly generated, timely ones, as far as the TCP spec is
> > concerned, unless you're concerned they're beings spoofed.
> 
> I'm perfectly sincere about it.  I think pointing out that the technique
> mitigates several possibilities emphasizes its value from an engineering
> perspective.
> 
> I can't argue with your perfectly correct statement that a compliant TCP
> will treat the spoofed, out of date, erroneously generated, or damaged
> packets as correct and reset the connection.  The argument is whether
> that behavior is useful or not.  
> 
> I think it's not, but YMMV.

I think I've shown above that the utility of the behavior is exclusively
motivated by authentication, and that other justifications don't have valid
support.

You can still think it's useful. I don't think it's not useful, so much as not
necessary if you're really concerned about security, and potentially dangerous
(unncesssarily complicated and IPR-burdened) if you aren't.

> > >> What does that tell you about the other end that you didn't know? What
> > >> purpose relevant to resetting a connection does synchronizing state
> > >> serve if not to authenticate the other end?
> > > 
> > > If the RST that triggered the ACK was from another connection, the
> > > TCPsecure exchange tells you that the other end of this connection has
> > > not reset.  
> > 
> > That presumes that either the RST is floating in the network more than 2
> > MSLs or that your end decided to accept a connection it shouldn't have
> > since it didn't correctly wait 2 MSLs from a previous connection.
> 
> Again, technically correct.  I think that addressing the eventuality that
> (among other things) MSL is incorrectly estimated or configured with an
> extra packet exchange is good engineering.
> 
> Yes, I know that an "engineering decision" is often difficult to tell
> from an "egregious hack."  I think that this design is the former.

Good engineering to update TCP to handle incorrect MSL estimation would be a
substantial piece of work. It would require reevaluating everything from SYN
exchanges to window offset validation. At the end of the day, if you don't
trust the MSL estimation, you need anti-replay. Yet another good reason to use
something like IPsec if this is what you're true concerns are, rather than a
patchwork of solutions that approximate anti-replay in specific cases.

> > If we're going to patch TCP to fix >2MSL errors, there's a lot more work
> > than this to be done. If your end is broken, then expecting this patch
> > to correct the situation is not productive.
> 
> Again, engineering vs. protocol design.
> 
> IMHO, we're already addressing unlikely events - MSL violation or
> off-path spoofed RST with known addresses and ports.  I think that the
> cost of one packet exchange to validate that the connection is not about
> to be terminated by an unlikely event is a reasonable engineering
> choice. 

TCP was designed to be able to unilaterally terminate a connection with a RST.
If you wanted a handshake, why did you not use FIN?

..
> > > No new information about the identity of that endpoint has
> > > been asserted, simply that the end that got your ACK has state closely
> > > enough synchronized with yours to make a sensical reply.  The identity
> > > of the sender of the RST remains unknown (though if the other end *has*
> > > reset, the sender is likely the other end).
> > 
> > The identity of the other end is defined by the synchronization process
> > - it's a TCP endpoint with which your ACKs synchronize state.
> > 
> > > I don't know anything new about the identity of anyone in the
> > > communication; I just don't see authentication there.
> > 
> > You know it's the other end with which you can synchronize state. That's
> > identity.
> 
> I think we're arguing over what to name the process.  This would be fun
> to do in person, but I don't think it's advancing the discussion of the
> system, or in particular of the discussion about the guidance to
> implementers we're trying to decide on.

I agree with that, but we've tripped over some other name issues that are
fundamental here:


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From: touch@ISI.EDU
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Subject: Re: [tcpm] tcpsecure: how strong to recommend?
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Quoting touch@ISI.EDU:

..
> > I think we're arguing over what to name the process.  This would be fun
> > to do in person, but I don't think it's advancing the discussion of the
> > system, or in particular of the discussion about the guidance to
> > implementers we're trying to decide on.
> 
> I agree with that, but we've tripped over some other name issues that are
> fundamental here:

, notably whether this is an update to RFC793 (I think most of us agree that it
is).

It's important for implementers to know why we're doing what we're doing, and
this is purely motivated by security concerns, and provides no real protection
from incorrect MSL estimation to TCP as a whole. That's why calling it
authentication is important. Calling it a patch to update TCP's robustness is
incorrect and misleads implementers into adopting this mechanism unncessarily.

Joe


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From: Ted Faber <faber@ISI.EDU>
To: touch@ISI.EDU
Subject: Re: [tcpm] tcpsecure: how strong to recommend?
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On Fri, Oct 05, 2007 at 10:21:38AM -0700, touch@ISI.EDU wrote:
> > IMHO, we're already addressing unlikely events - MSL violation or
> > off-path spoofed RST with known addresses and ports.  I think that the
> > cost of one packet exchange to validate that the connection is not about
> > to be terminated by an unlikely event is a reasonable engineering
> > choice.=20
>=20
> TCP was designed to be able to unilaterally terminate a connection with a=
 RST.
> If you wanted a handshake, why did you not use FIN?

That's an instructive example.

If your RST is lost - they're not sent reliably as a FIN is - the packet
exchange is exactly the one TCPsecure proposes: ACK the last packet.
(Strictly speaking the lost RST doesn't do it of course - that would be
a nice trick.  Either new data goes out (ACKing the last packet as
well), or a retransmission occurs, or a window probe goes off.)
Receiving the ACK triggers a retransmission of the RST and state
synchronizes.  If all the RSTs are lost, the state still synchronizes
with a timeout.  I wouldn't call that an authentication (and I know you
didn't), but the process is very similar to how TCPsecure plays out.

If we were in a drinking establishment or academic symposium where the
navel gazing were appropriate, we could discuss whether that RST packet
"unilaterally terminates the connection" or is a hint that the
connection has been unilaterally terminated by the destruction of the
connection state at one endpoint.   I'm sure you can tell what I
believe, and I'm suggesting that thinking of both these processes as
confirming a hint might be less objectionable to you than thinking of
them as authenticating a message.  It's a sadder world that we're not in
one of those places.

But here on the list, I'm hoping that talking about what a "slightly
unusual" packet is might steer us back toward the
MAY/SHOULD/MUST/applicability statement discussion.

--=20
Ted Faber
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Subject: Re: [tcpm] tcpsecure: how strong to recommend?
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On Fri, Oct 05, 2007 at 10:28:49AM -0700, touch@ISI.EDU wrote:
> Quoting touch@ISI.EDU:
>=20
> ..
> > > I think we're arguing over what to name the process.  This would be f=
un
> > > to do in person, but I don't think it's advancing the discussion of t=
he
> > > system, or in particular of the discussion about the guidance to
> > > implementers we're trying to decide on.
> >=20
> > I agree with that, but we've tripped over some other name issues that a=
re
> > fundamental here:
>=20
> , notably whether this is an update to RFC793 (I think most of us
> agree that it is).

Yep.  It's not much of an exercise to point at the paragraphs that
change.

>=20
> It's important for implementers to know why we're doing what we're doing,=
 and
> this is purely motivated by security concerns, and provides no real prote=
ction
> from incorrect MSL estimation to TCP as a whole. That's why calling it
> authentication is important. Calling it a patch to update TCP's robustnes=
s is
> incorrect and misleads implementers into adopting this mechanism unncessa=
rily.

I do think it makes TCP slightly more robust, but the occurrances that
would trigger that robustness are pretty unusual.  The most common cause of
that uncommon set is malice.  Do you think that concentrating on the
likelihood of bad RSTs (and SYNs and data) showing up would steer us
back toward the recommendation level?

--=20
Ted Faber
http://www.isi.edu/~faber           PGP: http://www.isi.edu/~faber/pubkeys.=
asc
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From: Ted Faber <faber@ISI.EDU>
To: touch@ISI.EDU
Subject: Re: [tcpm] tcpsecure: how strong to recommend?
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On Fri, Oct 05, 2007 at 11:39:31AM -0700, Ted Faber wrote:
> But here on the list, I'm hoping that talking about what a "slightly
> unusual" packet is might steer us back toward the
> MAY/SHOULD/MUST/applicability statement discussion.

Who the heck do I think I'm quoting there?

Sorry.

--=20
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Ted Faber wrote:
=2E..
> I do think it makes TCP slightly more robust, but the occurrances that
> would trigger that robustness are pretty unusual.  The most common caus=
e of
> that uncommon set is malice.  Do you think that concentrating on the
> likelihood of bad RSTs (and SYNs and data) showing up would steer us
> back toward the recommendation level?

I don't think considering these RSTs/SYNs/data issues more than
malicious is appropriate, and I do think that has a lot to do with the
recommendation level.

Joe


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Ted Faber wrote:
> On Fri, Oct 05, 2007 at 10:21:38AM -0700, touch@ISI.EDU wrote:
>>> IMHO, we're already addressing unlikely events - MSL violation or
>>> off-path spoofed RST with known addresses and ports.  I think that th=
e
>>> cost of one packet exchange to validate that the connection is not ab=
out
>>> to be terminated by an unlikely event is a reasonable engineering
>>> choice.=20
>> TCP was designed to be able to unilaterally terminate a connection wit=
h a RST.
>> If you wanted a handshake, why did you not use FIN?
>=20
> That's an instructive example.
>=20
> If your RST is lost - they're not sent reliably as a FIN is - the packe=
t
> exchange is exactly the one TCPsecure proposes: ACK the last packet.
> (Strictly speaking the lost RST doesn't do it of course - that would be=

> a nice trick.  Either new data goes out (ACKing the last packet as
> well), or a retransmission occurs, or a window probe goes off.)
>
> Receiving the ACK triggers a retransmission of the RST and state
> synchronizes.  If all the RSTs are lost, the state still synchronizes
> with a timeout.  I wouldn't call that an authentication (and I know you=

> didn't), but the process is very similar to how TCPsecure plays out.

It's not the same, as you point out. If there's no further data to be
sent, then it doesn't necessarily behave like a FIN exchange.

If you want a coordinated endpoint exchange, I suggest using a FIN.

> If we were in a drinking establishment or academic symposium where the
> navel gazing were appropriate, we could discuss whether that RST packet=

> "unilaterally terminates the connection" or is a hint that the
> connection has been unilaterally terminated by the destruction of the
> connection state at one endpoint.=20

I doubt we would debate that long; I agree with you, I suspect, that the
latter is more precise. It's a hint.

Again, a reason why you either take the hint or not. You don't respond
to it, and you don't gain anything - except over-optimization - by
telling the other end anything to confirm that hint.

> I'm sure you can tell what I
> believe, and I'm suggesting that thinking of both these processes as
> confirming a hint might be less objectionable to you than thinking of
> them as authenticating a message.=20

Confirming that it's a hint? Confirming what? That the hint came from
who you thought. Again we're back to authentication.

> But here on the list, I'm hoping that talking about what a "slightly
> unusual" packet is might steer us back toward the
> MAY/SHOULD/MUST/applicability statement discussion.

It walks, looks, and quacks like a duck. Whether you will agree to call
it a duck or not, I won't worry. But I've treated it like a duck in
deciding MUST/MAY/SHOULD.

If we truly want to get back to that discussion, I haven't heard others
address the proposed way forward I offered...

Joe


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Folks-

We are starting a WGLC on the UTO specification,
draft-ietf-tcpm-tcp-uto-06.txt.  We believe the document has addressed
the concerns raised by the WG and that there is good consensus for
publishing the document.  Our charter says the intended status for the
document is "Proposed Standard".  However, in the draft tracker it says
"Experimental".  We cannot seem to recall crisply deciding this issue.
So, please review the document one last time and raise issues on the
mailing list and please also weigh in on whether you think it should be
Experimental or Proposed Standard.  Also, notes of the form "looks fine
to me" are encouraged.  The WGLC will run through October 24.

Thanks,
Mark & Ted




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Generally the UTO draft seems absolutely fine and I would support it
moving to RFC. There seems no obvious reason for it to be experimental
rather than proposed standard.=20

I do have one potential concern and some nits:

Concern:

Section 4.1 Middleboxes suggests that "In the future, such [stateful]
firewalls may learn to parse the TCP User Timeout Option and adapt
connection state management accordingly." Would it be worth adding that
in this case this could become a potential security issue as it would
allow cooperative users to cause a stateful firewall to maintain
connection state for over 22 hours?

Nits:

Section 3:=20
'LOCAL_UTO' "If unspecified, it default to the default system-wide USER
TIMEOUT." change to "...defaults to the default..."

Para starting 'Before opening a connection...' "The default is to allow
this for connections that do not have a specific user timeout concerns."
delete "a"
"...to prevent UTO options from the other end to override local
application requests." change to "options from the other end overriding
local..."

Next para: "... is a reliable way to initially exchange and potentially
adapt to UTO values." add commas: "... is a reliable way to initially
exchange, and potentially adapt to, UTO values."

Section 3.1 para 1. Delete the first comma in the last sentence so it
reads: "In this case they SHOULD, however, notify the application about
the user timeout value received from the other end."

Para starting "This means that..." change last sentence (remove
nevertheless) "...it can still close or abort..."

Section 4.2 last sentence change to "Therefore, if a connection that
enables keep-alives is also using the TCP User Timeout Option, ..."

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

                                              =20
-----Original Message-----
From: Mark Allman [mailto:mallman@icir.org]=20
Sent: 10 October 2007 14:36
To: tcpm@ietf.org
Subject: [tcpm] WGLC for UTO

=20
Folks-

We are starting a WGLC on the UTO specification,
draft-ietf-tcpm-tcp-uto-06.txt.  We believe the document has addressed
the concerns raised by the WG and that there is good consensus for
publishing the document.  Our charter says the intended status for the
document is "Proposed Standard".  However, in the draft tracker it says
"Experimental".  We cannot seem to recall crisply deciding this issue.
So, please review the document one last time and raise issues on the
mailing list and please also weigh in on whether you think it should be
Experimental or Proposed Standard.  Also, notes of the form "looks fine
to me" are encouraged.  The WGLC will run through October 24.

Thanks,
Mark & Ted





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From tcpm-bounces@ietf.org Wed Oct 10 12:03:16 2007
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From: Lars Eggert <lars.eggert@nokia.com>
Subject: Re: [tcpm] WGLC for UTO
Date: Wed, 10 Oct 2007 10:59:42 -0500
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On 2007-10-10, at 10:22, ext tcpm-bounces@ietf.org wrote:
> Concern:
>
> Section 4.1 Middleboxes suggests that "In the future, such [stateful]
> firewalls may learn to parse the TCP User Timeout Option and adapt
> connection state management accordingly." Would it be worth adding  
> that
> in this case this could become a potential security issue as it would
> allow cooperative users to cause a stateful firewall to maintain
> connection state for over 22 hours?

Similar to how UTO is not binding for the ends (i.e., they may close  
a connection at any time), it is not binding for middleboxes. So  
these (so far purely hypothetical) middleboxes are free to time out  
the binding at any time, for example, if they become resource- 
constrained. We could make that explicit in the draft, if people feel  
the current text is unclear.

> Nits:

We'll fix these.

Thanks,
Lars
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Subject: Re: [tcpm] WGLC for UTO
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On Oct 10, 2007, at 16:36, ext Mark Allman wrote:

> We are starting a WGLC on the UTO specification,
> draft-ietf-tcpm-tcp-uto-06.txt.  We believe the document has addressed
> the concerns raised by the WG and that there is good consensus for
> publishing the document.  Our charter says the intended status for the
> document is "Proposed Standard".  However, in the draft tracker it  
> says
> "Experimental".  We cannot seem to recall crisply deciding this issue.

The draft looks fine to me to go forward.

Regarding the Proposed Standard vs. Experimental issue, I think this  
draft, like the other new TCP extensions, should first go to  
Experimental as a rule of thumb. I'm concerned about the increasing  
complexity and size of "the standards track" TCP implementation. It  
would be a good practice to have a careful evaluation in Experimental  
RFC phase before adding up things to standards track TCP that is  
expected to be implemented by most vendors.

- Pasi


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> The draft looks fine to me to go forward.

Many thanks for the note!

I have a couple questions (with my hat off) ...

> Regarding the Proposed Standard vs. Experimental issue, I think this
> draft, like the other new TCP extensions, should first go to
> Experimental as a rule of thumb. I'm concerned about the increasing
> complexity and size of "the standards track" TCP implementation. It
> would be a good practice to have a careful evaluation in Experimental
> RFC phase before adding up things to standards track TCP that is
> expected to be implemented by most vendors.

(1) What 'careful evaluation' do you think should be done during
    Experimental?

    Or, put another way, how would we know that it was OK for UTO to
    later move to PS?

    My own opinion here is that this draft specifies a way to share some
    information and that is pretty benign.  I can't see where it hurts
    anything if stacks want to do it.  It is advisory information if
    stacks want to ignore it.  So, what's the real harm here in putting
    it on the standards track?  Or, what would be the benefit of going
    experimental?  My own answer to both questions is "very little".

(2) While I sympathize with the goal of keeping things simple and
    avoiding needless complexity, I do not believe that just because we
    put something on the standards track we expect it to be implemented
    in every TCP implementation.  Perhaps that is flawed reading of
    these things.

Just my two bits.

allman




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From: Pasi Sarolahti <pasi.sarolahti@nokia.com>
Subject: Re: [tcpm] WGLC for UTO 
Date: Mon, 15 Oct 2007 18:31:13 +0300
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On Oct 15, 2007, at 14:48, ext Mark Allman wrote:

>> Regarding the Proposed Standard vs. Experimental issue, I think this
>> draft, like the other new TCP extensions, should first go to
>> Experimental as a rule of thumb. I'm concerned about the increasing
>> complexity and size of "the standards track" TCP implementation. It
>> would be a good practice to have a careful evaluation in Experimental
>> RFC phase before adding up things to standards track TCP that is
>> expected to be implemented by most vendors.

First of all, I want to clarify that I am proposing this as a general  
rule of thumb on how to introduce new features to TCP. It has been  
always a bit fuzzy to me when some new feature should go to  
experimental, and when it can go directly to standards track. This is  
not only about UTO (which I think seems like a fine scheme).

> (1) What 'careful evaluation' do you think should be done during
>     Experimental?

In case of UTO, given
* possibly limited budget of TCP implementation size (in embedded  
devices, for example)
* limited TCP option space
* possibility of resource exhaustion attacks as discussed in security  
considerations

Should I implement UTO and when should I enable it?

>     Or, put another way, how would we know that it was OK for UTO to
>     later move to PS?

If people find UTO useful (I'm sure many will do), they will  
implement UTO even as an Experimental RFC. After a couple of years of  
deployment experience advancing the document to PS should be a  
straight-forward process, if no problems have emerged, right?

One could ask the above question for any other experimental document.  
Having couple of years of kind of a "trial period" for new TCP  
extensions wouldn't do any harm, would it?

>     My own opinion here is that this draft specifies a way to share  
> some
>     information and that is pretty benign.  I can't see where it hurts
>     anything if stacks want to do it.  It is advisory information if
>     stacks want to ignore it.  So, what's the real harm here in  
> putting
>     it on the standards track?  Or, what would be the benefit of going
>     experimental?  My own answer to both questions is "very little".

I agree that UTO seems quite safe and non-controversial extension  
(like many other experimental RFC).

But one could also ask what's the real harm in putting it to  
experimental first? What would be the benefit of going directly to  
standards track, instead? Very little? Remembering a past tsvarea  
presentation at IETF-68 about Window Vista features, where Window  
scaling (PS) and ECN (PS) were disabled by default, and DSACK (Exp),  
F-RTO (Exp) enabled...

I realize that in the end it is up to the implementors to decide  
which standards track RFCs they choose to use, and we have indeed  
seen that some implementations are pretty liberal in selecting the  
TCP features. Nevertheless, it would be useful to have a meaning to  
the standards track status, indicating that a feature is considered  
to be part of the "recommended TCP", to help the people pondering  
among the number of TCP RFCs out there. The roadmap RFC seems to make  
this assumption, too, having the standards track enhancements listed  
as "recommended".

- Pasi


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From tcpm-bounces@ietf.org Mon Oct 15 12:16:53 2007
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Subject: Re: [tcpm] WGLC for UTO 
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[[ Folks, we could use some opinions from additional people on this 
   question. ]]

> First of all, I want to clarify that I am proposing this as a general
> rule of thumb on how to introduce new features to TCP. It has been
> always a bit fuzzy to me when some new feature should go to
> experimental, and when it can go directly to standards track. This is
> not only about UTO (which I think seems like a fine scheme).

Point taken.

> > (1) What 'careful evaluation' do you think should be done during
> >     Experimental?
> 
> In case of UTO, given
> * possibly limited budget of TCP implementation size (in embedded
> * devices, for example)
> * limited TCP option space
> * possibility of resource exhaustion attacks as discussed in
> * security  considerations

The thing that I don't see is how we'll differently answer these
questions in 2 or 3 years than we do today.  I don't mean to claim there
are not considerations.  There are.  But, are embedded devices that
don't want needless TCP mechanisms going to be gone in a few years?  How
is that consideration going to change with experience?  In a few years
will we be able to discount resource attacks?

It seems to me that you (and others) are advocating experience for the
sake of experience, rather than experience for the sake of figuring
something in particular out.

> One could ask the above question for any other experimental
> document.

Sure---and I think we should.  But, some documents have good answers and
some don't, IMO.

E.g., we made NCR an experimental RFC because it delays TCP's congestion
response and therefore makes TCP more aggressive.  I think the question
"is this OK?" is clear and would have to be answered with experimental
evidence before we decided that it was OK for standards track (i.e.,
generally applicable for wide spread deployment).  In this case "is it
OK?" is a technical question about the mechanism.  I think this is in
direct contrast to UTO---which is about the exchange of information.  We
can say we're going to ask "is it OK?", but in the UTO case that is not
a technical question.  How do we answer that question with an
experiment?  I don't see any reason why we'd delay putting UTO on the
standards track as a **proposed** standard.

> But one could also ask what's the real harm in putting it to
> experimental first? What would be the benefit of going directly to
> standards track, instead? Very little?

Point taken.  The benefit may also be small.

In some sense it depends on the default setting you like best.  You want
to add an extra step to all documents and I want to add that extra step
only when it seems like we really need to add that step (and, it doesn't
seem to me that we need the extra step in this case).

> Remembering a past tsvarea
> presentation at IETF-68 about Window Vista features, where Window
> scaling (PS) and ECN (PS) were disabled by default, and DSACK (Exp),
> F-RTO (Exp) enabled...

(Nit: DSACK is PS and always has been.  And, IMO this is important
because it is quite close to UTO because they both specify how to share
information and not much else.)

allman




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From tcpm-bounces@ietf.org Tue Oct 16 14:42:16 2007
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Pasi Sarolahti wrote:
>=20
> On Oct 15, 2007, at 14:48, ext Mark Allman wrote:
>=20
>>> Regarding the Proposed Standard vs. Experimental issue, I think this
>>> draft, like the other new TCP extensions, should first go to
>>> Experimental as a rule of thumb.=20

I don't agree. Whether something goes to experimental is determined by
whether there is an experiment needed. The standards track has phases
for us to gain confidence with extensions - that's why things don't just
go straight to Standard.

There are three choices, BTW:

	- standards track
		should be for things we think are useful to the
		Internet as a whole, and not harmful to the Internet
		as a whole if widely deployed

	- experimental
		things for which an experiment in deployment will help
		either gain confidence in group behavior or parameter
		determination

	- informational
		things which are not harmful to the Internet as a
		whole if widely deployed, but may be useful in
		specific environments

Anything that is harmful to the Internet as a whole should be documented
more as something to avoid, IMO.

>>> I'm concerned about the increasing
>>> complexity and size of "the standards track" TCP implementation.=20

That should be a consideration for all items in the standards track, and
is largely - AFAICT - determined by the MUST/SHOULD/MAY designations.

I will note, as others have as well, that there's a problem with the
menu of choices out there. Many choices are intermingled, and the
roadmap should be clear on which current choices are related. The same
relationship can be true for items within a mechanism, as we've discused
on this list as well.

Perhaps it would be useful for us to adopt an informal requirement that
TCP mods come with three clear summaries:

1) whether the proposed mechanism is, as a whole, MUST/SHOULD/MAY

2) which components of the proposed mechanism are MUST/SHOULD/MAY

3) what other dependent choices are MUST/SHOULD/MAY

It'd be useful to have a section that lists these succinctly in summary,
at the beginning of the document, e.g., "Requirements Levels and
Dependencies".

>> (1) What 'careful evaluation' do you think should be done during
>>     Experimental?
>=20
> In case of UTO, given
> * possibly limited budget of TCP implementation size (in embedded
> devices, for example)

MAY and SHOULD allow designers to omit things for space constraints.

> * limited TCP option space

MAY and SHOULD allow designers - and users - to disable things for
header space limitations.

> * possibility of resource exhaustion attacks as discussed in security
> considerations

That ought to be addressed before being released. Security questions are
not the kinds of things we want to gain experience with via widescale
deployment.

> Should I implement UTO and when should I enable it?

MAY handles that.

Joe


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Just a reminder that we have one week left on thr UTO WGLC.  We have one
comment so far.  Please take a few minutes and give this one more read
and send a note to the list with your thoughts.

Thanks,
allman





> Folks-
> 
> We are starting a WGLC on the UTO specification,
> draft-ietf-tcpm-tcp-uto-06.txt.  We believe the document has addressed
> the concerns raised by the WG and that there is good consensus for
> publishing the document.  Our charter says the intended status for the
> document is "Proposed Standard".  However, in the draft tracker it says
> "Experimental".  We cannot seem to recall crisply deciding this issue.
> So, please review the document one last time and raise issues on the
> mailing list and please also weigh in on whether you think it should be
> Experimental or Proposed Standard.  Also, notes of the form "looks fine
> to me" are encouraged.  The WGLC will run through October 24.
> 
> Thanks,
> Mark & Ted




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From: Pasi Sarolahti <pasi.sarolahti@nokia.com>
Subject: Re: [tcpm] WGLC for UTO 
Date: Thu, 18 Oct 2007 17:01:29 +0300
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On Oct 15, 2007, at 19:13, ext Mark Allman wrote:

> E.g., we made NCR an experimental RFC because it delays TCP's  
> congestion
> response and therefore makes TCP more aggressive.  I think the  
> question
> "is this OK?" is clear and would have to be answered with experimental
> evidence before we decided that it was OK for standards track (i.e.,
> generally applicable for wide spread deployment).  In this case "is it
> OK?" is a technical question about the mechanism.  I think this is in
> direct contrast to UTO---which is about the exchange of  
> information.  We
> can say we're going to ask "is it OK?", but in the UTO case that is  
> not
> a technical question.  How do we answer that question with an
> experiment?  I don't see any reason why we'd delay putting UTO on the
> standards track as a **proposed** standard.

Ok, I think you are right that it is hard to come up with well- 
defined experiments that would give some additional technical  
information about the UTO. But I don't understand where this  
requirement for experiments comes from. I also ask, because this  
"what is the experiment?" question has come up quite a few times  
during the past couple of years for Experimental documents in  
different WGs, and RFC 2026 does not say anything about experiments:

    The "Experimental" designation typically denotes a specification  
that
    is part of some research or development effort.  Such a  
specification
    is published for the general information of the Internet technical
    community and as an archival record of the work, subject only to
    editorial considerations and to verification that there has been
    adequate coordination with the standards process (see below)....

(I believe many of the proposed TCP extensions are part of a research  
project.)

> In some sense it depends on the default setting you like best.  You  
> want
> to add an extra step to all documents and I want to add that extra  
> step
> only when it seems like we really need to add that step (and, it  
> doesn't
> seem to me that we need the extra step in this case).

I didn't mean adding extra step to *all* documents, just that we  
should by default be conservative when extending standards track TCP  
with new features. The idea was that during the Experimental state we  
could, among other things, get better understanding whether there is  
"enough community interest to be considered valuable", as required  
for Proposed Standard. This is sometimes hard to judge based on the  
(lack of) comments during WGLCs. In addition to the comments on the  
mailing list, a good indication of "community interest" would be, if  
someone indicated interest in taking a documented extension into a  
real implementation.

>> Remembering a past tsvarea
>> presentation at IETF-68 about Window Vista features, where Window
>> scaling (PS) and ECN (PS) were disabled by default, and DSACK (Exp),
>> F-RTO (Exp) enabled...
>
> (Nit: DSACK is PS and always has been.  And, IMO this is important
> because it is quite close to UTO because they both specify how to  
> share
> information and not much else.)

Sorry, by "DSACK" I meant Experimental RFC 3708, i.e. the usage part.  
The UTO document is different that RFC 2883 because it specifies both  
the new option, and recommended response on receiving the option.

- Pasi


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From tcpm-bounces@ietf.org Thu Oct 18 10:20:35 2007
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Pasi Sarolahti wrote:
> Ok, I think you are right that it is hard to come up with well-defined
> experiments that would give some additional technical information about=

> the UTO. But I don't understand where this requirement for experiments
> comes from. I also ask, because this "what is the experiment?" question=

> has come up quite a few times during the past couple of years for
> Experimental documents in different WGs, and RFC 2026 does not say
> anything about experiments:
>=20
>    The "Experimental" designation typically denotes a specification tha=
t
>    is part of some research or development effort.  Such a specificatio=
n
>    is published for the general information of the Internet technical
>    community and as an archival record of the work, subject only to
>    editorial considerations and to verification that there has been
>    adequate coordination with the standards process (see below)....
>=20
> (I believe many of the proposed TCP extensions are part of a research
> project.)

That has rarely been the case; more typically, the 'research' is being
conducted by the proposer and the Internet community at large to
determine the efficacy of a spec, and the 'experiment' is part of that
'research or development effort'.

Joe


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From tcpm-bounces@ietf.org Thu Oct 18 13:38:03 2007
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Subject: RE: [tcpm] WGLC for UTO 
Date: Thu, 18 Oct 2007 13:36:07 -0400
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UTO manages to provide value for those who want it with
no adverse impact on existing implementations. It is also
easy for stacks (and stack offloads) to add support.

So I fully support allowing this to proceed to standard.



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I had a few minor points; I don't think any are showstoppers, but the WG
might consider whether it'd be useful to fix these before this goes out
the door. I think it should, since some are holes in behavior that ought
to be defined.

---

- MUST/MAY/SHOULD on the whole option
	I didn't see this; I'm not sure I care what the decision is,
	but it should be stated in the abstract and intro

- sec 3 at "Before opening..." for two paragraphs
This section uses 2119 language in ways I don't understand. An app that
wishes to do something SHOULD? If it wishes to do something, it "DOES"
it. I don't see a reason for 2119 there, or MAY at the end of that
paragraph, or MAY in the next. In these cases, there are choices
involved that determine what happens. You cannot dictated userland /
application behavior when defining these events in this document, IMO.

- sec 3 "In addition to"
There is no reason in requiring (via SHOULD) a refresh of
SYN-coordinated state. Either that state is coordinated, or it is not.
If not, then the connection is in error.

- sec 3 "A host that supports"
What happens if there is no option space in the next outgoing segment?
What happens if there is no option space at all? Do you tell the app? Do
you do anything special? IMO, this deserves a separate section rather
than just a note at the end of 3.0

- sec 3.1 "When a host"
"In this case, they SHOULD" - they who?

- sec 3.1 "In general"
This is contrary to the default values. CHANGEABLE defaults to true, but
ENABLED defaults to false. That means, IMO, that received UTO options
would be dropped, not acted upon, in the default case.

- sec 6
I thought this section should say a few specific things:

1) There are no new IANA namespaces. (which you say)

2) This doc requires IANA to assign a value from the TCP Option
namespace, to be referred in section 3.3 accordingly. (which you don't
quite ask for!)

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

Very minor points, if an edit occurs:

- intro near "applications can set"
apps typically set/change this param via the OPEN/SEND socket calls, not
the TCP OPEN/SEND API. The two use the same names; it'd be useful to be
clear here.

- intro/throughout
'other end' seems informal; other end of the TCP connection, etc.?

- intro last two paragraphs
these would be useful to move earlier

- sec 3
'per-application basis' -> 'per-connection basis'

- sec 3 at LOCAL_UTO
it should be more clear what the relationship between the system-wide
USER TIMEOUT and local USER_TIMEOUT is; alternately, a list of the
global vars would be useful.

- sec 3.1 "This means that"
This paragraph is part of a discussion on checks and limits; this should
not be the first place you introduce the "max" concept.

- hysteresis?
The doc doesn't describe how quickly the UTO can be changed by the
remote end or application. Does it matter, e.g., for security reasons?

- sec 3.2 "The host requirements rfc"
"seconds is RECOMMENDED" -> append "in that RFC" (you're not doing the
recommending; you're quoting 1122)

- sec 3.2
If you retransmit the option and attach it to a particular byte, and the
byte is ACKd, then why don't you KNOW the option has been received
(i.e., second part of the handshake)? if the other end receives data
over 1 window away from that byte, why doesn't it KNOW the source end
knows it has been recieved (i.e., third part of the handshake)?

About retransmission, IMO, it SHOULD be done - or else what's the point
of this?

- sec 3.4
"TCP implementations" -> "Current TCP implementations"

- sec 4.1
Firewalls may parse this option in unencrypted TCP segments only.

- sec 5
in addition to IPsec, cite RFC2385 for peer authentication


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Date: Fri, 19 Oct 2007 19:35:34 -0700
From: "Vishal Study" <vishal.study@gmail.com>
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Subject: [tcpm] SYN in SYN_RCVD state
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Consider the following scenario:

1. TCP client sends SYN to server; server goes to SYN_SENT state

2. server responds with SYN+ACK;  server goes to SYN_RCVD state
but SYN+ACK sent to client is lost somewhere in the network.

3. client re-sends SYN on its timeout (couple of seconds later) to server

What should be the server behavior? Should it retransmit SYN+ACK or
should it send a RST?

thanks!


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Subject: [tcpm] New  I-D
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A new version of an I-D has been posted on the IETF web site.

http://www.ietf.org/internet-drafts/draft-mahesh-persist-timeout-02.txt
"TCP Maintenance and Minor Extensions", Mahesh Jethanandani, Murali Bashyam, 27-Oct-07, <draft-mahesh-persist-timeout-02.txt> 


Abstract

   This document describes how a connection can remain infinitely in
   persist condition, and its Denial of Service (DoS) implication on the
   system, if there is no mechanism to recover from this anomaly.

Summary:
    In this version of the draft, we have documented our experiment 
using a simple user level program to create the DoS scenario. The tests 
were run against both Apache and IIS HTTP servers. The test was run on a 
larger scale in the lab environment, where we were able to document the 
behavior of the servers. The test was then run against public and well 
known sites but on a smaller scale. Sniffer traces were captured to see 
the behavior of the connections.

Our observations were that of the three well known public sites 
exhibited the DoS scenario. While site A had some mitigation technique 
in place, it was fairly easy to beat that. Site B and C had no 
mitigation in place and are currently vulnerable to the DoS attack.

Solution:
The draft documents some suggested solutions and describes why they are 
better than any of the techniques out there. In particular it talks 
about UTO and why UTO cannot solve this problem. It also talks about the 
role of applications and how they can help.

Comments are welcome.

/mahesh


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From: "Vishal Study" <vishal.study@gmail.com>
To: "Marco Mellia" <mellia@tlc.polito.it>
Subject: Re: [tcpm] SYN in SYN_RCVD state
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But RFC 793, pg 71 seem to say that the server should send out a RST
if SYN is rx in SYN-RCVD state (assuming SYN seq number is within the
window, which is true in the example I had mentioned).

Any thoughts on this?

Thanks!

On 10/20/07, Marco Mellia <mellia@tlc.polito.it> wrote:
> Vishal Study ha scritto:
> > Consider the following scenario:
> >
> > 1. TCP client sends SYN to server; server goes to SYN_SENT state
> >
> > 2. server responds with SYN+ACK;  server goes to SYN_RCVD state
> > but SYN+ACK sent to client is lost somewhere in the network.
> >
> > 3. client re-sends SYN on its timeout (couple of seconds later) to server
> >
> > What should be the server behavior? Should it retransmit SYN+ACK or
> > should it send a RST?
> >
> >
> Clearly the first case!
> Marco
>


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On October 21, 2007, Vishal wrote -

> But RFC 793, pg 71 seem to say that the server should send out a RST
> if SYN is rx in SYN-RCVD state (assuming SYN seq number is within the
> window, which is true in the example I had mentioned).
>

Your example is covered by the rules on page 68 of RFC 793. The
retransmitted SYN segment is a duplicate segment and is outside
the receive window (SEG.SEQ is < RCV.NXT).=20
Hence, the receiver sends an Acknowledgement and discards the segment.

Anil

Anil Agarwal
ViaSat Inc.
Germantown, MD 20876
USA


On 10/20/07, Marco Mellia <mellia@tlc.polito.it> wrote:
> Vishal Study ha scritto:
> > Consider the following scenario:
> >
> > 1. TCP client sends SYN to server; server goes to SYN_SENT state
> >
> > 2. server responds with SYN+ACK;  server goes to SYN_RCVD state
> > but SYN+ACK sent to client is lost somewhere in the network.
> >
> > 3. client re-sends SYN on its timeout (couple of seconds later) to
server
> >
> > What should be the server behavior? Should it retransmit SYN+ACK or
> > should it send a RST?
> >
> >
> Clearly the first case!
> Marco
>


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To: Vishal Study <vishal.study@gmail.com>
Subject: Re: [tcpm] SYN in SYN_RCVD state
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On Sun, Oct 21, 2007 at 04:25:25AM -0700, Vishal Study wrote:
> > Vishal Study ha scritto:
> > > Consider the following scenario:
> > >
> > > 1. TCP client sends SYN to server; server goes to SYN_SENT state
> > >
> > > 2. server responds with SYN+ACK;  server goes to SYN_RCVD state
> > > but SYN+ACK sent to client is lost somewhere in the network.
> > >
> > > 3. client re-sends SYN on its timeout (couple of seconds later) to se=
rver
> > >
> > > What should be the server behavior? Should it retransmit SYN+ACK or
> > > should it send a RST?
> > >
> But RFC 793, pg 71 seem to say that the server should send out a RST
> if SYN is rx in SYN-RCVD state (assuming SYN seq number is within the
> window, which is true in the example I had mentioned).
>=20
> Any thoughts on this?

I think that because RCV.NEXT was set to SEG.SEQ+1 when the "server"
entered SYN-RECEIVED state (as documented on page 66 of 793) the resent
SYN is outside the window and will be acked.  The "server's" SYN+ACK
will be resent if it times out.

--=20
Ted Faber
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From: "Vishal Study" <vishal.study@gmail.com>
To: "Ted Faber" <faber@isi.edu>
Subject: Re: [tcpm] SYN in SYN_RCVD state
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Hi Ted:

The server would retransmit SYN+ACK when server times out in SYN-RECEIVED state.
I agree and have seen this.

But, when the server hasn't timed out in SYN-RECEIVED state and it
receives retransmitted SYN from client (due to client's timeout), what
is the expected behavior on server side? Is server supposed to send
SYN+ACK?

Thanks!

On 10/22/07, Ted Faber <faber@isi.edu> wrote:
> On Sun, Oct 21, 2007 at 04:25:25AM -0700, Vishal Study wrote:
> > > Vishal Study ha scritto:
> > > > Consider the following scenario:
> > > >
> > > > 1. TCP client sends SYN to server; server goes to SYN_SENT state
> > > >
> > > > 2. server responds with SYN+ACK;  server goes to SYN_RCVD state
> > > > but SYN+ACK sent to client is lost somewhere in the network.
> > > >
> > > > 3. client re-sends SYN on its timeout (couple of seconds later) to server
> > > >
> > > > What should be the server behavior? Should it retransmit SYN+ACK or
> > > > should it send a RST?
> > > >
> > But RFC 793, pg 71 seem to say that the server should send out a RST
> > if SYN is rx in SYN-RCVD state (assuming SYN seq number is within the
> > window, which is true in the example I had mentioned).
> >
> > Any thoughts on this?
>
> I think that because RCV.NEXT was set to SEG.SEQ+1 when the "server"
> entered SYN-RECEIVED state (as documented on page 66 of 793) the resent
> SYN is outside the window and will be acked.  The "server's" SYN+ACK
> will be resent if it times out.
>
> --
> Ted Faber
> http://www.isi.edu/~faber           PGP: http://www.isi.edu/~faber/pubkeys.asc
> Unexpected attachment on this mail? See http://www.isi.edu/~faber/FAQ.html#SIG
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>


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On Mon, Oct 22, 2007 at 12:37:23PM -0700, Vishal Study wrote:
> Hi Ted:
>=20
> The server would retransmit SYN+ACK when server times out in SYN-RECEIVED=
 state.
> I agree and have seen this.
>=20
> But, when the server hasn't timed out in SYN-RECEIVED state and it
> receives retransmitted SYN from client (due to client's timeout), what
> is the expected behavior on server side? Is server supposed to send
> SYN+ACK?

The spec says no.   RFC793 p 69 gives the sequence number test and the
ACK requirement:

----
   first check sequence number

      SYN-RECEIVED STATE
      ESTABLISHED STATE
      FIN-WAIT-1 STATE
      FIN-WAIT-2 STATE
      CLOSE-WAIT STATE
      CLOSING STATE
      LAST-ACK STATE
      TIME-WAIT STATE

        Segments are processed in sequence.  Initial tests on arrival
        are used to discard old duplicates, but further processing is
        done in SEG.SEQ order.  If a segment's contents straddle the
        boundary between old and new, only the new parts should be
        processed.

[table and some text snipped.  None of it applies; the segment is
outside the window and unacceptable. ]

        If an incoming segment is not acceptable, an acknowledgment
        should be sent in reply (unless the RST bit is set, if so drop
        the segment and return):

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

        After sending the acknowledgment, drop the unacceptable segment
        and return.
----

I can imagine an implementor sending a SYN+ACK here, but it would be
outside the spec.  I assume you have no basis for sending one.

--=20
Ted Faber
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Ted Faber wrote:
> On Mon, Oct 22, 2007 at 12:37:23PM -0700, Vishal Study wrote:
>> Hi Ted:
>>
>> The server would retransmit SYN+ACK when server times out in SYN-RECEI=
VED state.
>> I agree and have seen this.
>>
>> But, when the server hasn't timed out in SYN-RECEIVED state and it
>> receives retransmitted SYN from client (due to client's timeout), what=

>> is the expected behavior on server side? Is server supposed to send
>> SYN+ACK?
>=20
> The spec says no.=20

Agreed. FWIW, Ted already said there are two separate events:

1) the arriving retransmitted SYN causes the server to send an ACK

2) a timeout on the server associated with sending the SYN-ACK goes off,
and the server resends the SYN-ACK then

(1) doesn't cause (2); timeout does.

Joe


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I endorse Joe Touch's comments of the 18th of Oct.
I am in favor, with those changes, of advancing UTO.

Brian Dickson


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Folks,

We have documented a case of HTTP servers that are prone to resource starvation with the use of a small user level program. The program does not require any special privileges or changes in the kernel. The user level program on the client opens a connection to a HTTP server, sends a GET request for a large file (larger than the advertised window of the client) but never reads the response.

Three well-known, public sites were tested for this vulnerability. The two most common HTTP servers, Apache and IIS were the target. While one site had put mitigation technique in place, the others had none. With the latter two we were able to hold connections in ESTABLISHED state for days. The former site had a mitigation in place with a fixed timeout of 11 min., which was easy to guess and work around.

We (the authors) believe that this is a huge problem. What do you folks feel?


Previous responses to this documentation has been that it is a application problem. It is clear from our experimentation that most HTTP servers (and FTP too) have not implemented any mitigation techniques. We believe that this problem exists across the whole range of TCP based applications prevalent on the internet, although our experiments were limited to the web application. Where applications have tried to put mitigation techniques in place, workaround has been easy. This is mainly because applications do not have the same amount of visibility as TCP does on the state of the connection.

If you are interested in reading the draft, you will find it here <http://www.ietf.org/internet-drafts/draft-mahesh-persist-timeout-02.txt>.

/mahesh


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<pre wrap="">Folks,

We have documented a case of HTTP servers that are prone to resource starvation with the use of a small user level program. The program does not require any special privileges or changes in the kernel. The user level program on the client opens a connection to a HTTP server, sends a GET request for a large file (larger than the advertised window of the client) but never reads the response.

<span class="moz-txt-citetags"></span>Three well-known, public sites were tested for this vulnerability. The two most common HTTP servers, Apache and IIS were the target. While one site had put mitigation technique in place, the others had none. With the latter two we were able to hold connections in ESTABLISHED state for days. The former site had a mitigation in place with a fixed timeout of 11 min., which was easy to guess and work around.

We (the authors) believe that this is a huge problem. What do you folks feel?
</pre>
<pre wrap=""><!---->
Previous responses to this documentation has been that it is a application problem. It is clear from our experimentation that most HTTP servers (and FTP too) have not implemented any mitigation techniques. We believe that this problem exists across the whole range of TCP based applications prevalent on the internet, although our experiments were limited to the web application. Where applications have tried to put mitigation techniques in place, workaround has been easy. This is mainly because applications do not have the same amount of visibility as TCP does on the state of the connection.

If you are interested in reading the draft, you will find it <a
 href="http://www.ietf.org/internet-drafts/draft-mahesh-persist-timeout-02.txt">here</a>.

/mahesh
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From tcpm-bounces@ietf.org Mon Oct 29 18:04:58 2007
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This is indeed a problem. I'm just not convinced it's a transport
problem.

=20

Only the application layer has enough information to know when a
client's

behavior is abusive and/or suspicious. The transport layer should remain

militantly ignorant of these matters.

=20


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<p class=3DMsoNormal><span =
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color:#1F497D'>This is indeed a problem. I&#8217;m just not convinced =
it&#8217;s a
transport problem.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Only the application layer has enough information to know =
when a
client&#8217;s<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>behavior is abusive and/or suspicious. The transport =
layer
should remain<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>militantly ignorant of these =
matters.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

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Mahesh Jethanandani wrote:
> Folks,
>=20
> We have documented a case of HTTP servers that are prone to resource
> starvation with the use of a small user level program. The program does=

> not require any special privileges or changes in the kernel. The user
> level program on the client opens a connection to a HTTP server, sends =
a
> GET request for a large file (larger than the advertised window of the
> client) but never reads the response.
>=20
> Three well-known, public sites were tested for this vulnerability.=20
> The two most common HTTP servers, Apache and IIS were the target.
> While one site had put mitigation technique in place, the others had
> none. With the latter two we were able to hold connections in
> ESTABLISHED state for days. The former site had a mitigation in place
> with a fixed timeout of 11 min., which was easy to guess and work
> around.
>=20
> We (the authors) believe that this is a huge problem. What do you
> folks feel?

I agree with Caitlin and your previous responses; this is an application
problem.

> Previous responses to this documentation has been that it is a
> application problem. It is clear from our experimentation that most HTT=
P
> servers (and FTP too) have not implemented any mitigation techniques. W=
e
> believe that this problem exists across the whole range of TCP based
> applications prevalent on the internet, although our experiments were
> limited to the web application. Where applications have tried to put
> mitigation techniques in place, workaround has been easy. This is mainl=
y
> because applications do not have the same amount of visibility as TCP
> does on the state of the connection.

Applications know when a connection hasn't closed. They also know when a
write would block due to insufficient socket resources. Either or both
of these provide sufficient visibility to avoid the problem entirely.

As you note, applications can solve this easily. Since it's their
obligation to do so, and it's easy for them to do so, there's no reason
to complicate the transport layer with this responsibility.

Furthermore, does the propose solution solve the DOS problem? Aren't
there other ways to keep a source stalled? (i.e., what about continued
SACKs? or repeated ACKs indicating a lost segment? at the very least, we
could ACK a byte at a time, sending 3-4 duplicate ACKs for each byte,
which would keep the sender from opening its window and stall things a
lot...).

Joe


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Caitlin,

Caitlin Bestler wrote:
>
> This is indeed a problem. I'm just not convinced it's a transport problem.
>
>  
>
> Only the application layer has enough information to know when a client's
>
> behavior is abusive and/or suspicious.
>
Applications if anything have less knowledge than TCP on why a 
connection is not making progress. In this particular case, HTTP could 
not make a distinction between a connection not making progress because 
of congestion in the network or because the client is advertising zero 
window. And yet all Apache can do is to reset the connection after a 
fixed amount of time. Again, HTTP may know how many of its connections 
are in the stuck state, it will not know how many FTP connections are 
seeing the same issue or how seriously constrained the system is.

We have found that a fixed timeout also does not work. It is easy to 
guess and work around as we did with our client program.

The solution has to be adaptive to the condition which in this case 
involves TCP and applications have little knowledge of the state of all 
of TCP connections.
>
> The transport layer should remain
>
> militantly ignorant of these matters.
>
When it impacts TCP's ability to function, I can hardly see that being 
ignorant will help.

--------------090700010209010108080505
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Caitlin,<br>
<br>
Caitlin Bestler wrote:
<blockquote cite="mid:78C9135A3D2ECE4B8162EBDCE82CAD7702765252@nekter"
 type="cite">
  <meta http-equiv="Content-Type" content="text/html; ">
  <meta name="Generator" content="Microsoft Word 12 (filtered medium)">
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  <p class="MsoNormal"><span
 style="font-size: 11pt; font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">This
is indeed a problem. I&#8217;m just not convinced it&#8217;s a
transport problem.<o:p></o:p></span></p>
  <p class="MsoNormal"><span
 style="font-size: 11pt; font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);"><o:p>&nbsp;</o:p></span></p>
  <p class="MsoNormal"><span
 style="font-size: 11pt; font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">Only
the application layer has enough information to know when a
client&#8217;s<o:p></o:p></span></p>
  <p class="MsoNormal"><span
 style="font-size: 11pt; font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">behavior
is abusive and/or suspicious. </span></p>
  </div>
</blockquote>
Applications if anything have less knowledge than TCP on why a
connection is not making progress. In this particular case, HTTP could
not make a distinction between a connection not making progress because
of congestion in the network or because the client is advertising zero
window. And yet all Apache can do is to reset the connection after a
fixed amount of time. Again, HTTP may know how many of its connections
are in the stuck state, it will not know how many FTP connections
are seeing the same issue or how seriously constrained the system is. <br>
<br>
We have found that a fixed timeout also does not work. It is easy to
guess and work around as we did with our client program. <br>
<br>
The solution has to be adaptive to the condition which in this case
involves TCP and applications have little knowledge of the state of all
of TCP connections.<br>
<blockquote cite="mid:78C9135A3D2ECE4B8162EBDCE82CAD7702765252@nekter"
 type="cite">
  <div class="Section1">
  <p class="MsoNormal"><span
 style="font-size: 11pt; font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">The
transport layer
should remain<o:p></o:p></span></p>
  <p class="MsoNormal"><span
 style="font-size: 11pt; font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31, 73, 125);">militantly
ignorant of these matters.</span></p>
  </div>
</blockquote>
When it impacts TCP's ability to function, I can hardly see that being
ignorant will help.<br>
</body>
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From tcpm-bounces@ietf.org Tue Oct 30 01:19:52 2007
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Mahesh Jethanandani wrote:
> Caitlin,
>=20
> Caitlin Bestler wrote:
>>
>> This is indeed a problem. I=92m just not convinced it=92s a transport =
problem.
>> Only the application layer has enough information to know when a clien=
t=92s
>> behavior is abusive and/or suspicious.
>>
> Applications if anything have less knowledge than TCP on why a
> connection is not making progress. In this particular case, HTTP could
> not make a distinction between a connection not making progress because=

> of congestion in the network or because the client is advertising zero
> window.=20

If an application - or host - has limited resources, it doesn't matter
why a connection isn't making progress.

> And yet all Apache can do is to reset the connection after a
> fixed amount of time. Again, HTTP may know how many of its connections
> are in the stuck state, it will not know how many FTP connections are
> seeing the same issue or how seriously constrained the system is.

There is no rule that TCP connections "see" each other per se; from
within a single connection, an implementation could be able to access
only a single connection's state.

And, ultimately, a system is constrained when it runs out of processor
or memory - which have nothing to do necessarily with the connections in
progress. I.e., an application can ask the OS - or the OS can tell the
app - when such limitations come into play anyway.

> We have found that a fixed timeout also does not work. It is easy to
> guess and work around as we did with our client program.
>=20
> The solution has to be adaptive to the condition which in this case
> involves TCP and applications have little knowledge of the state of all=

> of TCP connections.
>>
>> The transport layer should remain
>> militantly ignorant of these matters.
>>
> When it impacts TCP's ability to function, I can hardly see that being
> ignorant will help.

You're declaring that someone who sends a zero receive window longer
than you expect to be noncompliant. There's nothing in the spec that
says that a TCP connection has to make forward progress in any fixed
timescale - even based on RTTs. This has allowed TCP to proceed over
shared links where less than one segment succeeds per RTT - e.g.,
congested transatlantic links. My concern is that this modification will
disable correct behavior in the corner cases.

Not all TCP connections must - or should - work only for the common cases=
=2E

Joe


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MURALI BASHYAM wrote:
> I agree with Caitlin and your previous responses; this is an
>  application
> problem.
>=20
> So we are placing the responsibility for protecting system-wide TCP res=
ources such as buffers
> and connections on all the TCP based applications out there?=20

Yes - applications can always start extra TCP connections, and can
override default socket sizes, so those resources are already under
application control. They are managed by the OS as a whole, i.e., when
the OS runs out, the application gets that feedback, and reacts according=
ly.

> Which "application"
> are you referring to here? BGP? NFS ? CIFS? HTTP? SMTP? FTP? The entire=
 application
> community? Potentially several or all of these apps are vulnerable. How=
 do we bring all these=20
> protocols into a single agreeable framework and  accomplish the right t=
hing?=20

Any application can misbehave. Any application can hold off receiving
data it has asked for. If the sender side application cares, it cuts off
the connection. If it doesn't, it leaves the connection in place.

TCP doesn't read minds.

=2E..
> Applications cannot discriminate between a connection that's stalled du=
e to network
> congestion and a connection that's being stalled by a malicious receive=
r who has closed
> his receive window forever. And for how long the connection has been in=
 in this stalled state etc.

Applications shouldn't discriminate between these. First, there's no
difference between a connection that's being stalled maliciously and one
that's being stalled legitimately. There's no requirement that TCP not
be stalled for a particular interval. You'd be disabling TCP's ability
(feature) to work over such environments arbitrarily if you change this.

> And what about multi-application servers? Surely each application here =
is running as an
> independent process and indvidually they don't have the system wide awa=
reness that TCP has...

Nor should it. First, as I noted in other mail, TCP doesn't necessarily
have system-wide information - that's an *assumption*. You're talking
about the kernel making decisions, not TCP. TCP could be implemented in
user space, or in the kernel in a way that it is isolated from
information about other connections. That's another reason this isn't
TCP's job.

> As you note, applications can solve this easily. Since it's their
> obligation to do so, and it's easy for them to do so, there's no reason=

> to complicate the transport layer with this responsibility.
>=20
> Applications can implement a fixed or even an adaptive timeout, but unl=
ess they can make
> the above distinction i am pointing out, they will be knocking out legi=
timate (albeit slow)=20
> connections, i.e there will be a fairness issue.=20

And they'll still be knocking out legitimate (albeit slow) receivers
that have a good reason to send a zero window for some period of time -
like while you insert a new DVD to backup to.

> Applications do not have visibility into TCP connection
> sender's state to solve the fairness issue.=20

Visibility into the state of a single connection does not solve this
issue. You simply know that the receiver is not accepting data at this
time - this is a feature of TCP. It's not a bug.

> A solution which penalizes legitimate connections is clearly a broken o=
ne.

Agreed!

> Furthermore, does the propose solution solve the DOS problem? Aren't
> there other ways to keep a source stalled? (i.e., what about continued
> SACKs? or repeated ACKs indicating a lost segment? at the very least,
>  we
> could ACK a byte at a time, sending 3-4 duplicate ACKs for each byte,
> which would keep the sender from opening its window and stall things a
> lot...).

> How do you propose to carry out these attacks without changing the clie=
nt side TCP stack in the=20
> windows or linux kernel? The vulnerability which we have demonstrated a=
nd addressed requires a simple user=20
> level program which anyone with a little, basic programming background =
can write today. Clearly the bar to creating these attacks you are pointi=
ng out is higher.

It's not a vulnerability unless the application lets it be one. It's a
feature, and no, it shouldn't be disabled just because one
interpretation of what is happening is "DOS attack".

Is it a DOS attack to receive lots of connections at once? Is it a DOS
attack to recieve a connection that opens then doesn't send data for a
while? All of these are legitimate behaviors of TCP - as is having the
receiver stall the sender.

Joe


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Subject: Re: [tcpm] Is this a problem?
To: Joe Touch <touch@ISI.EDU>, Mahesh Jethanandani <mahesh@cisco.com>
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----- Original Message ----
From: Joe Touch <touch@ISI.EDU>
To: Mahesh Jethanandani <mahesh@cisco.com>
Cc: tcpm@ietf.org
Sent: Monday, October 29, 2007 3:42:41 PM
Subject: Re: [tcpm] Is this a problem?




Mahesh Jethanandani wrote:
> Folks,
> 
> We have documented a case of HTTP servers that are prone to resource
> starvation with the use of a small user level program. The program
 does
> not require any special privileges or changes in the kernel. The user
> level program on the client opens a connection to a HTTP server,
 sends a
> GET request for a large file (larger than the advertised window of
 the
> client) but never reads the response.
> 
> Three well-known, public sites were tested for this vulnerability. 
> The two most common HTTP servers, Apache and IIS were the target.
> While one site had put mitigation technique in place, the others had
> none. With the latter two we were able to hold connections in
> ESTABLISHED state for days. The former site had a mitigation in place
> with a fixed timeout of 11 min., which was easy to guess and work
> around.
> 
> We (the authors) believe that this is a huge problem. What do you
> folks feel?

I agree with Caitlin and your previous responses; this is an
 application
problem.

So we are placing the responsibility for protecting system-wide TCP resources such as buffers
and connections on all the TCP based applications out there? Which "application"
are you referring to here? BGP? NFS ? CIFS? HTTP? SMTP? FTP? The entire application
community? Potentially several or all of these apps are vulnerable. How do we bring all these 
protocols into a single agreeable framework and  accomplish the right thing? 

> Previous responses to this documentation has been that it is a
> application problem. It is clear from our experimentation that most
 HTTP
> servers (and FTP too) have not implemented any mitigation techniques.
 We
> believe that this problem exists across the whole range of TCP based
> applications prevalent on the internet, although our experiments were
> limited to the web application. Where applications have tried to put
> mitigation techniques in place, workaround has been easy. This is
 mainly
> because applications do not have the same amount of visibility as TCP
> does on the state of the connection.

Applications know when a connection hasn't closed. They also know when
 a
write would block due to insufficient socket resources. Either or both
of these provide sufficient visibility to avoid the problem entirely.

Applications cannot discriminate between a connection that's stalled due to network
congestion and a connection that's being stalled by a malicious receiver who has closed
his receive window forever. And for how long the connection has been in in this stalled state etc.

And what about multi-application servers? Surely each application here is running as an
independent process and indvidually they don't have the system wide awareness that TCP has...

As you note, applications can solve this easily. Since it's their
obligation to do so, and it's easy for them to do so, there's no reason
to complicate the transport layer with this responsibility.

Applications can implement a fixed or even an adaptive timeout, but unless they can make
the above distinction i am pointing out, they will be knocking out legitimate (albeit slow) 
connections, i.e there will be a fairness issue. Applications do not have visibility into TCP connection
sender's state to solve the fairness issue. A solution which penalizes legitimate connections is clearly a broken one.

Furthermore, does the propose solution solve the DOS problem? Aren't
there other ways to keep a source stalled? (i.e., what about continued
SACKs? or repeated ACKs indicating a lost segment? at the very least,
 we
could ACK a byte at a time, sending 3-4 duplicate ACKs for each byte,
which would keep the sender from opening its window and stall things a
lot...).

How do you propose to carry out these attacks without changing the client side TCP stack in the 
windows or linux kernel? The vulnerability which we have demonstrated and addressed requires a simple user 
level program which anyone with a little, basic programming background can write today. Clearly the bar to creating these attacks you are pointing out is higher.




Joe





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Joe Touch wrote:
> Applications know when a connection hasn't closed. They also know when a
> write would block due to insufficient socket resources. Either or both
> of these provide sufficient visibility to avoid the problem entirely.
>   
Knowing those two facts only gives applications some of the information. 
It still would not know why the write is blocked? You agree that 
penalizing legitimate connections is a broken implementation. How does 
application then differentiate between a legitimate connection that is 
slow and one that is malicious (by advertising a zero window for an 
extended period of time)?

It is important to understand that a slow legitimate connection is a 
behavior that is almost entirely contained within TCP but the client 
sending zero window can be a case of a malicious user level program 
(read externally triggered attack) not reading data. Looking into the 
TCP connection state allows us to differentiate between the two 
different conditions and treat the connections fairly.
> As you note, applications can solve this easily. Since it's their
> obligation to do so, and it's easy for them to do so, there's no reason
> to complicate the transport layer with this responsibility.
>   
The reality is that few applications if any implement any mechanism to 
detect a connection that is stalled in such a manner. Our experiment 
shows that  the most commonly used application (HTTP) used on some of 
the biggest web sites are vulnerable today holding on to connections in 
EST state for days. Yes, they should implement a mechanism, but can TCP 
count on it. What is it supposed to do if applications do not clear 
connections in a timely manner.

> Furthermore, does the propose solution solve the DOS problem? Aren't
> there other ways to keep a source stalled? (i.e., what about continued
> SACKs? or repeated ACKs indicating a lost segment? at the very least, we
> could ACK a byte at a time, sending 3-4 duplicate ACKs for each byte,
> which would keep the sender from opening its window and stall things a
> lot...).
>   
You are talking about a malicious TCP implementation that would require 
special permissions in a DDoS scenario. What we used was a simple user 
level program that required no special privileges to run.


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From tcpm-bounces@ietf.org Tue Oct 30 09:35:15 2007
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Mahesh Jethanandani wrote:
>=20
>=20
> Joe Touch wrote:
>> Applications know when a connection hasn't closed. They also know when=
 a
>> write would block due to insufficient socket resources. Either or both=

>> of these provide sufficient visibility to avoid the problem entirely.
>>  =20
> Knowing those two facts only gives applications some of the information=
=2E
> It still would not know why the write is blocked? You agree that
> penalizing legitimate connections is a broken implementation.=20

My point is that your solution does this.

> How does
> application then differentiate between a legitimate connection that is
> slow and one that is malicious (by advertising a zero window for an
> extended period of time)?

Since it never can, it shouldn't try. It depends on the application.
That's why it's the application's decision as to when to give up on a
client, not TCP's.

> It is important to understand that a slow legitimate connection is a
> behavior that is almost entirely contained within TCP but the client
> sending zero window can be a case of a malicious user level program
> (read externally triggered attack) not reading data.=20

I agree that sending a zero window CAN be an attack. Unfortunately, it
CAN be legitimate behavior.

Since you agree we can't decide between the two at the TCP layer, we
should not fix this at that layer either.

> Looking into the
> TCP connection state allows us to differentiate between the two
> different conditions and treat the connections fairly.

You are asserting that "all zero-window receivers are attackers". This
is incorrect, and leading you to the conclusion that TCP should fix this.=


A legitimate receiver could have a backed-up application - i.e., a slow
processor, more pressing other applications, etc. - which could cause it
to advertise a zero window for an arbitrary time.

There is no requirement of timeliness in TCP data processing. When data
arrives at the receiver, if it backs up, the receiver could send a zero
window. That condition could legitimately persist a long time - I gave a
real example before (a backup device waiting for a human to insert a new
DVD).

>> As you note, applications can solve this easily. Since it's their
>> obligation to do so, and it's easy for them to do so, there's no reaso=
n
>> to complicate the transport layer with this responsibility.
>>  =20
> The reality is that few applications if any implement any mechanism to
> detect a connection that is stalled in such a manner.

I agree. That doesn't make it TCP's responsibility.

> Our experiment
> shows that  the most commonly used application (HTTP) used on some of
> the biggest web sites are vulnerable today holding on to connections in=

> EST state for days. Yes, they should implement a mechanism, but can TCP=

> count on it. What is it supposed to do if applications do not clear
> connections in a timely manner.

TCP is the servant of the application, not its master. If the
application doesn't clear connections TCP should - and must - sit and wai=
t.

>> Furthermore, does the propose solution solve the DOS problem? Aren't
>> there other ways to keep a source stalled? (i.e., what about continued=

>> SACKs? or repeated ACKs indicating a lost segment? at the very least, =
we
>> could ACK a byte at a time, sending 3-4 duplicate ACKs for each byte,
>> which would keep the sender from opening its window and stall things a=

>> lot...).
>>  =20
> You are talking about a malicious TCP implementation that would require=

> special permissions in a DDoS scenario. What we used was a simple user
> level program that required no special privileges to run.

Yes, but as I've shown, your case is indistinguishable from legitimate
behavior. Let's not 'fix' that.

Joe


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From: Florian Weimer <fw@deneb.enyo.de>
To: Mahesh Jethanandani <mahesh@cisco.com>
Subject: Re: [tcpm] Is this a problem?
References: <472654F9.5030308@cisco.com>
Date: Tue, 30 Oct 2007 15:14:19 +0100
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* Mahesh Jethanandani:

> Three well-known, public sites were tested for this vulnerability. The
> two most common HTTP servers, Apache and IIS were the target. While
> one site had put mitigation technique in place, the others had
> none. With the latter two we were able to hold connections in
> ESTABLISHED state for days. The former site had a mitigation in place
> with a fixed timeout of 11 min., which was easy to guess and work
> around.
>
> We (the authors) believe that this is a huge problem. What do you
> folks feel?

It's a problem, but not at the TCP layer.

SSH connections, for instance, can legitimately last for days in
ESTABLISHED state without any traffic being exchanged.

For HTTP, it may be helpful to have some kind of information regarding
the TCP state to mitigate the DoS potential, but I fear that it's too
easily manipulated by the attacker to be useful.  Even if it's not, it's
really a sockets API issue.


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From: Lars Eggert <lars.eggert@nokia.com>
Subject: Re: [tcpm] WGLC for UTO
Date: Tue, 30 Oct 2007 17:31:21 +0200
To: ext Joe Touch <touch@ISI.EDU>
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On 2007-10-19, at 2:10, ext Joe Touch wrote:
> - MUST/MAY/SHOULD on the whole option
> 	I didn't see this; I'm not sure I care what the decision is,
> 	but it should be stated in the abstract and intro

I'd like to ask the opinion of the WG here: Should we recommend that  
all stacks implement UTO as a PS? Or only some stacks, and in this  
case, which ones?

(Alternatively, if we're going for Experimental with this document,  
this question is moot, because implicitly only hosts desiring to  
experiment with this option need to implement it.)

> - sec 3 at "Before opening..." for two paragraphs
> This section uses 2119 language in ways I don't understand. An app  
> that
> wishes to do something SHOULD? If it wishes to do something, it "DOES"
> it. I don't see a reason for 2119 there, or MAY at the end of that
> paragraph, or MAY in the next. In these cases, there are choices
> involved that determine what happens. You cannot dictated userland /
> application behavior when defining these events in this document, IMO.

I've rephrased these sentences to not use 211 terms.

> - sec 3 "In addition to"
> There is no reason in requiring (via SHOULD) a refresh of
> SYN-coordinated state. Either that state is coordinated, or it is not.
> If not, then the connection is in error.

The intent was to make UTO work with SYN cookie implementations that  
can't encode whether a UTO option was present in the cookie, by  
including the option also in the first non-SYN packets. It may also  
help when the option space in the SYN packets was too full to allow  
UTO to be included. So there is some use to doing this. We could  
state that including UTo in the first non-SYN packets is only useful  
if the option was not included in the SYN?

> - sec 3 "A host that supports"
> What happens if there is no option space in the next outgoing segment?
> What happens if there is no option space at all? Do you tell the  
> app? Do
> you do anything special? IMO, this deserves a separate section rather
> than just a note at the end of 3.0

I don't know what we can say about this - this is a general issue  
with TCP. What should be done depends a lot on which other options  
are in use for a particular connection. Maybe UTO is considered more  
important than some options, but less important than others. Which  
options do you include and which ones do you not include, generally?  
Hard to say.

> - sec 3.1 "When a host"
> "In this case, they SHOULD" - they who?

s/they/TCP/ - fixed

> - sec 3.1 "In general"
> This is contrary to the default values. CHANGEABLE defaults to  
> true, but
> ENABLED defaults to false. That means, IMO, that received UTO options
> would be dropped, not acted upon, in the default case.

When the draft was targeted as Experimental, having ENABLED default  
to false was needed, so that hosts would need to actively enable the  
mechanism to participate to experiment with the option.

If we're going with PS (are we?), ENABLED should default to true, I  
guess.

> - sec 6
> I thought this section should say a few specific things:
>
> 1) There are no new IANA namespaces. (which you say)
>
> 2) This doc requires IANA to assign a value from the TCP Option
> namespace, to be referred in section 3.3 accordingly. (which you don't
> quite ask for!)

Right - I made this explicit.

> Very minor points, if an edit occurs:
...

Most of these are addressed in the next revision. But not these, so far:

> - sec 3.1 "This means that"
> This paragraph is part of a discussion on checks and limits; this  
> should
> not be the first place you introduce the "max" concept.

It's not - the formula further up introduces it.

> - hysteresis?
> The doc doesn't describe how quickly the UTO can be changed by the
> remote end or application. Does it matter, e.g., for security reasons?

I don't think there are security implications here. The only result  
of changing the value very frequently is that all segments have the  
option attached, so there is some mild increase in bandwidth use. But  
the app can always just send more data anyway :-)

> - sec 3.2
> If you retransmit the option and attach it to a particular byte,  
> and the
> byte is ACKd, then why don't you KNOW the option has been received
> (i.e., second part of the handshake)? if the other end receives data
> over 1 window away from that byte, why doesn't it KNOW the source end
> knows it has been recieved (i.e., third part of the handshake)?

*If* you do this, yes, but there was consensus earlier that the WG  
didn't want the UTO exchange to be reliable, which is why this  
"attaching" the option to a byte is a MAY.

> About retransmission, IMO, it SHOULD be done - or else what's the  
> point
> of this?

Is this related to your previous comment? If so, there was consensus  
to not make UTO reliable, i.e., retransmit lost options.

Thanks!

Lars


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On 2007-10-29, at 19:33, ext Alfred H=CEnes wrote:
> (1)  Terminology -- Section 3, 3.1 et al.
>
> Section 3 newly has introduced the variable 'USER_TIMEOUT':
>
>    USER_TIMEOUT
>       TCP's USER TIMEOUT parameter, as specified in [RFC0793].
>
> The subsequent text uses 'user timeout', 'USER TIMEOUT', and
> 'USER_TIMEOUT' in a way that seems to be a bit confusing; the
> distinction between these terms, and the rationale for selecting
> one of these terms remains unclear in some places.

USER TIMEOUT refers to the RFC793-defined TCP state variable.

USER_TIMEOUT names a variable in the formula in Section 3.1. that is =20
used to set the local USER TIMEOUT. It's not a TCP state variable; =20
USER TIMEOUT is.

The lower-case phrase is used in the body of the text when the =20
concept is talked about, and not one of the specific items above. At =20
least that was the intent - we may have used the lower case in some =20
places where we should have used the upper case form.

> Also, the title of Section 3.1 contains 'User Timeout' whereas most
> occurrences of 'user timeout' have been changed to 'USER TIMEOUT'
> in the body of Section 3.1.

All section titles have initial capitals.

> I propose to reconsider the distinction, and to use 'USER TIMEOUT'
> only in the immediate context of RFC 793 as opposed to this draft;
> most of the occurrences of 'USER TIMEOUT' should better be replaced
> by 'USER_TIMEOUT' (the TCP state variable) or 'user timeout' (the
> abstract concept as seen from the application / user).

USER_TIMEOUT isn't the TCP state variable, USER TIMEOUT (from 793) is.

I'm OK with using the lower-case variant throughout the text, if =20
people think that would eliminate confusion?

> Also, the names of the two variables, USER_TIMEOUT and LOCAL_UTO,
> are not good mnemonics for the intended purpose of these variables;
> IMHO, the naming should be reconsidered again (I already had argued
> in this matter).  In particular, 'LOCAL_UTO' now has much diminuished
> *local* significance, it it just the value advertised to the peer
> (=3D> my new proposal: ADV_UTO ); USER_TIMEOUT ist the important local
> state variable, and there's now only a loose coupling between both.

I like ADV_UTO, change made.

> (2)  Section 3
> In the second-to-last paragraph of Section 3, there's an unnecessary
> word replication:

fixed

> (3)  Section 3.3
> In the last paragraph, remove the duplicate full-stop in =20
> "connection.."

done

Thanks!

Lars=

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Subject: Re: [tcpm] WGLC for UTO
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Lars Eggert wrote:
> On 2007-10-19, at 2:10, ext Joe Touch wrote:
=2E..
>> - sec 3 "In addition to"
>> There is no reason in requiring (via SHOULD) a refresh of
>> SYN-coordinated state. Either that state is coordinated, or it is not.=

>> If not, then the connection is in error.
>=20
> The intent was to make UTO work with SYN cookie implementations that
> can't encode whether a UTO option was present in the cookie, by
> including the option also in the first non-SYN packets.=20

That should be deemed non-compliant. The point of the SYN cookie is to
reestablish the entirety of state that would have been established by
the SYN; if the cookie can't do that, a cookie should not be used.

TCP has no provisions for post-SYN establishment of new connection state
AFAIK. Requiring this of this option would be new, and, IMO, a bit
confusing for designers. What happens if the post-SYN packet lacks the
option? do you declare state mis-synchronization and drop it?

> It may also help
> when the option space in the SYN packets was too full to allow UTO to b=
e
> included.

IMO, if that's a concern, then make sure the UTO MUST NOT be included in
the SYN. I.e., it's in one place or the other; once in place, it
shouldn't be repeated in subsequent segments - only replayed in segments
being retransmitted.

> So there is some use to doing this. We could state that
> including UTo in the first non-SYN packets is only useful if the option=

> was not included in the SYN?

I'd prefer that, adjusted to say "if the option was not included in the
SYN due to space limitations".

>> - sec 3 "A host that supports"
>> What happens if there is no option space in the next outgoing segment?=

>> What happens if there is no option space at all? Do you tell the app? =
Do
>> you do anything special? IMO, this deserves a separate section rather
>> than just a note at the end of 3.0
>=20
> I don't know what we can say about this - this is a general issue with
> TCP. What should be done depends a lot on which other options are in us=
e
> for a particular connection. Maybe UTO is considered more important tha=
n
> some options, but less important than others. Which options do you
> include and which ones do you not include, generally? Hard to say.

I'm not making an assertion as to what to do, only that the doc ought to
consider the issue. I'll suggest that it ought to send it within 2
segments OR indicate to the application that it has failed; the latter
requires a new API variable, though.

>> - sec 3.1 "In general"
>> This is contrary to the default values. CHANGEABLE defaults to true, b=
ut
>> ENABLED defaults to false. That means, IMO, that received UTO options
>> would be dropped, not acted upon, in the default case.
>=20
> When the draft was targeted as Experimental, having ENABLED default to
> false was needed, so that hosts would need to actively enable the
> mechanism to participate to experiment with the option.
>=20
> If we're going with PS (are we?), ENABLED should default to true, I gue=
ss.

Sure - making the two consistent is the issue.

=2E..
>> Very minor points, if an edit occurs:
> ...
>=20
> Most of these are addressed in the next revision. But not these, so far=
:
>=20
>> - sec 3.1 "This means that"
>> This paragraph is part of a discussion on checks and limits; this shou=
ld
>> not be the first place you introduce the "max" concept.
>=20
> It's not - the formula further up introduces it.

It might be useful to introduce the issue in prose first. The reason for
using max is the issue, and it's not clear by just stating the formula.

>> - hysteresis?
>> The doc doesn't describe how quickly the UTO can be changed by the
>> remote end or application. Does it matter, e.g., for security reasons?=

>=20
> I don't think there are security implications here. The only result of
> changing the value very frequently is that all segments have the option=

> attached, so there is some mild increase in bandwidth use. But the app
> can always just send more data anyway :-)

Can someone on the path do something you don't want to your connection
by tacking this option on?

>> - sec 3.2
>> If you retransmit the option and attach it to a particular byte, and t=
he
>> byte is ACKd, then why don't you KNOW the option has been received
>> (i.e., second part of the handshake)? if the other end receives data
>> over 1 window away from that byte, why doesn't it KNOW the source end
>> knows it has been recieved (i.e., third part of the handshake)?
>=20
> *If* you do this, yes, but there was consensus earlier that the WG
> didn't want the UTO exchange to be reliable, which is why this
> "attaching" the option to a byte is a MAY.

Ah - it might be useful to explain that. I thought MAY implied that the
information about reliability was in doubt, rather than that the use of
the attachment was optional but would provide reliable UTO exchange if us=
ed.

>> About retransmission, IMO, it SHOULD be done - or else what's the poin=
t
>> of this?
>=20
> Is this related to your previous comment? If so, there was consensus to=

> not make UTO reliable, i.e., retransmit lost options.

I have no idea what the point of an unreliable option on a reliable
transport protocol means. IMO, options should be retransmitted when
their associated segments are. The control protocol ought to be at least
as robust as the data. Although I appreciate that UTO isn't as critical
as some other options or flags, introducing nondeterminism at this level
doesn't seem appropriate, IMO.

I.e., I think there are separate questions:

	1) is UTO critical enough to ensure correct exchange via
	an explicitly ack'd handshake
		no - I agree with the WG

	2) should options associated with a segment be retransmitted
	with that segment in general in TCP
		I think the answer to this should be YES, always,
		but I don't think it was asked.

Joe


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-----Original Message-----
From: Joe Touch [mailto:touch@ISI.EDU]=20
Sent: Tuesday, October 30, 2007 9:08 AM
To: Lars Eggert
Cc: tcpm@ietf.org
Subject: Re: [tcpm] WGLC for UTO



Lars Eggert wrote:
> On 2007-10-19, at 2:10, ext Joe Touch wrote:
...
>> - sec 3 "In addition to"
>> There is no reason in requiring (via SHOULD) a refresh of=20
>> SYN-coordinated state. Either that state is coordinated, or it is
not.
>> If not, then the connection is in error.
>=20
> The intent was to make UTO work with SYN cookie implementations that=20
> can't encode whether a UTO option was present in the cookie, by=20
> including the option also in the first non-SYN packets.

That should be deemed non-compliant. The point of the SYN cookie is to
reestablish the entirety of state that would have been established by
the SYN; if the cookie can't do that, a cookie should not be used.

TCP has no provisions for post-SYN establishment of new connection state
AFAIK. Requiring this of this option would be new, and, IMO, a bit
confusing for designers. What happens if the post-SYN packet lacks the
option? do you declare state mis-synchronization and drop it?

----------End of Original Message
--------------------------------------------------------

The idea here is to recognize that existing stacks establish the TCB at
two different times: on first SYN or on the Ack of the SYN-ACK.

To be very pedantic, the TCB is *always* established when the SYN is
received. It's just that a SYN Cookie encodes the "TCB" in the SYN
cookie value and "stores" the TCB on the network.

Which would be a problem with UTO if the UTO option were only present
in the SYN cookie. Basically there isn't any more room in the SYN
Cookie without stealing information from elsewhere (such as timing
resolution or cryptographic quality).

That won't happen. Instead anyone using SYN Cookies will simply NEVER
support UTO for *any* connection (in which case whether it was in the
SYN is irrelevant, which does encode very efficiently).

The redundant transmission of the UTO option allows a receiver that
uses SYN Cookies to establish connections safely to enable that support
only when it is *actually* creating the full-sized TCB. This same
benefit
actually applies to any strategy where allocation of a full TCB is
deferred
until after the round-trip (which is a generally recommended practice,
although there can be debate about how much the proto-TCB should be
condensed and where it should be stored).


As to Lars comment, the idea of having it both places is to allow=20
implementations that create the full TCB on receipt of the SYN to
properly create the TCB. Currently, most software based SYN Cookies
implementations include a "no attack detected" mode where the TCBs
are created immediately.



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Caitlin Bestler wrote:
>=20
> -----Original Message-----
> From: Joe Touch [mailto:touch@ISI.EDU]=20
> Sent: Tuesday, October 30, 2007 9:08 AM
> To: Lars Eggert
> Cc: tcpm@ietf.org
> Subject: Re: [tcpm] WGLC for UTO
>=20
> Lars Eggert wrote:
>> On 2007-10-19, at 2:10, ext Joe Touch wrote:
> ...
>>> - sec 3 "In addition to"
>>> There is no reason in requiring (via SHOULD) a refresh of=20
>>> SYN-coordinated state. Either that state is coordinated, or it is
> not.
>>> If not, then the connection is in error.
>> The intent was to make UTO work with SYN cookie implementations that=20
>> can't encode whether a UTO option was present in the cookie, by=20
>> including the option also in the first non-SYN packets.
>=20
> That should be deemed non-compliant. The point of the SYN cookie is to
> reestablish the entirety of state that would have been established by
> the SYN; if the cookie can't do that, a cookie should not be used.
>=20
> TCP has no provisions for post-SYN establishment of new connection stat=
e
> AFAIK. Requiring this of this option would be new, and, IMO, a bit
> confusing for designers. What happens if the post-SYN packet lacks the
> option? do you declare state mis-synchronization and drop it?
>=20
> ----------End of Original Message
> --------------------------------------------------------
>=20
> The idea here is to recognize that existing stacks establish the TCB at=

> two different times: on first SYN or on the Ack of the SYN-ACK.
>=20
> To be very pedantic, the TCB is *always* established when the SYN is
> received. It's just that a SYN Cookie encodes the "TCB" in the SYN
> cookie value and "stores" the TCB on the network.
>=20
> Which would be a problem with UTO if the UTO option were only present
> in the SYN cookie. Basically there isn't any more room in the SYN
> Cookie without stealing information from elsewhere (such as timing
> resolution or cryptographic quality).

TCP with SYN cookies must decline connections whose state it cannot
sufficiently encode in a cookie.

> That won't happen. Instead anyone using SYN Cookies will simply NEVER
> support UTO for *any* connection (in which case whether it was in the
> SYN is irrelevant, which does encode very efficiently).

That's inappropriate - the end that ACKs a SYN is responsible for making
sure the appropriate TCB is established. AFAIK, we don't have a signal
for the client to know that cookies are used and to retransmit options
on the next segment, so this would break ALL options established without
ACK in a SYN.

That may require UTO never be sent in a SYN at all. I don't see why that
would be a large problem, and it's cleaner than creating a situation
where a cookie doesn't restore full state.

(i.e., presuming UTO the only option in a SYN that can succeed
unacknowledged?)

> The redundant transmission of the UTO option allows a receiver that
> uses SYN Cookies to establish connections safely to enable that support=

> only when it is *actually* creating the full-sized TCB. This same
> benefit
> actually applies to any strategy where allocation of a full TCB is
> deferred
> until after the round-trip (which is a generally recommended practice,
> although there can be debate about how much the proto-TCB should be
> condensed and where it should be stored).

The TCB needs to encode enough information to establish the state that
the SYN-ACK explicitly agrees to. It's not up to the endpoint to decide
not to keep some of that state - once it's ACK'd, it's ACKd.

> As to Lars comment, the idea of having it both places is to allow=20
> implementations that create the full TCB on receipt of the SYN to
> properly create the TCB. Currently, most software based SYN Cookies
> implementations include a "no attack detected" mode where the TCBs
> are created immediately.

If this is the only option that has this property (silent source
assertion of a SYN option), then the appropriate solution is to prohibit
it from SYNs, IMO.

Joe


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From tcpm-bounces@ietf.org Tue Oct 30 15:00:48 2007
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Mahesh Jethanandani wrote:
> Folks,
> 
> We have documented a case of HTTP servers that are prone to resource starvation with the use of a small user level program. The program does not require any special privileges or changes in the kernel. The user level program on the client opens a connection to a HTTP server, sends a GET request for a large file (larger than the advertised window of the client) but never reads the response.

This problem has been well documented for at least seven years.
http://shlang.com/netkill/
http://shlang.com/netkill/20000421-netkill-bugtraq-announcement.txt


> Three well-known, public sites were tested for this vulnerability. The two most common HTTP servers, Apache and IIS were the target. While one site had put mitigation technique in place, the others had none. With the latter two we were able to hold connections in ESTABLISHED state for days. The former site had a mitigation in place with a fixed timeout of 11 min., which was easy to guess and work around.
> 
> We (the authors) believe that this is a huge problem. What do you folks feel?

I personally believe it is a problem, but the absence of wide-spread 
attacks using this technique leads me to believe that other attacks are 
still more effective.  All the same, I would like to see more widespread 
defense against it.


> Previous responses to this documentation has been that it is a application problem. It is clear from our experimentation that most HTTP servers (and FTP too) have not implemented any mitigation techniques. We believe that this problem exists across the whole range of TCP based applications prevalent on the internet, although our experiments were limited to the web application. Where applications have tried to put mitigation techniques in place, workaround has been easy. This is mainly because applications do not have the same amount of visibility as TCP does on the state of the connection.

I think nearly all responses (including mine) have been of the opinion 
that this is not a transport problem.  However, I think this is not 
necessarily an application problem, either.  Since the contended 
resource is system memory, the best place to implement a fairness policy 
is in the operating system -- possibly but not necessarily in the kernel 
(for OS's where such a distinction applies).

   -John


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Subject: RE: [tcpm] WGLC for UTO
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Joe Touch wrote:

.

> If this is the only option that has this property (silent source
> assertion of a SYN option), then the appropriate solution is to
> prohibit it from SYNs, IMO.

> Joe

If you're allocated a "full" TCB based on only an unconfirmed SYN
then you probably would not mind allocating one that is UTO capable.

So I don't really see any harm in prohibiting it from the SYN, unless
it's presence there is needed to keep connection tracking middleboxes
happy or somesuch. The passive endpoint certainly does not need it
to be in the SYN. Any timeouts before the connection is fully
established
are going to have their own logic anyway.





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Attached is a slightly tweaked version of the 2581 implementation
report.  The report includes input from the linux community noting that
2581 is implemented in their stack and that they have not seen any sort
of big problems because of it.

allman



 

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Background:

  + RFC 2581 is a re-write of RFC 2001.  RFC 2001 was a description
    of TCP's congestion control algorithms that was published long
    after these algorithms were in nearly ubiquitous deployment
    throughout the Internet (largely triggered by the congestion
    collapses of the mid-1980s).

  + While RFC 2001 was a description of the algorithms, RFC 2581 is
    a more traditional specification.  We stress that the RFC was
    written based on running code and experience.

  + The mechanisms in RFC 3042 (Limited Transmit) and RFC 3390
    (Larger Initial Congestion Window) are also rolled into the
    current document.  Both of these enhancements are Proposed
    Standards that have gathered wide consensus within the community
    based on deployment experience.

  + The traditional test of two interoperable implementations to
    move a Proposed Standard to Draft Standard is less obvious in
    the case of congestion control mechanisms.  Congestion control
    is about *when* to send a segment and not *what* that segment
    looks like, how to process it, how big fields are, etc.
    Therefore, it is difficult to assess "interoperability" in the
    traditional sense.  Below we cite several sources that show or
    suggest that multiple implementations of the mechanisms exist
    and seem to work as intended.

  + The new version of the document clarifies a number of small
    issues that implementers have asked about over the years, but
    does not make any large changes to the algorithms.

Known Implementations:

  + [WS95] discusses the BSD implementation of the core algorithms
    in RFC 2581 (slow start, congestion avoidance, fast retransmit
    and fast recovery).  This implementation has formed the basis of
    the TCP stack in numerous operating systems (NetBSD, FreeBSD,
    OpenBSD, SunOS 4.x, BSDI, etc.).  While various operating
    systems may have diverged in small details (some of which is
    documented in RFC 2581) the basic algorithms do not seem to have
    changed.

  + Linux also supports RFC 2581 and does not report any adverse
    impacts.  See Attachment 1 below.

    (The complaint in that email is not about the document itself or
    even the algorithm within RFC 2581, but rather goes to our
    congestion control principles.  Further, as sketched the
    behavior given in RFC 2581 is more conservative than desired and
    therefore if this RFC is in error, it is erroring in the right
    direction for stable operation.)

  + [Pax97] analyzes a number of implementations, finding both
    correct and incorrect behavior relative to RFC 2581 across a
    variety of implementations.  The incorrect behavior fed into
    [RFC2525].

  + [MAF05] tests for conformance along a number of angles by
    probing the TCPs of over 70K web servers with specialized packet
    streams that induce the stack to show how it handles various
    situations. The results include:

      + The vast majority of server reduce their congestion window
        by half in response to congestion (per RFC 2581's congestion
        avoidance).

      + The majority of the web servers used an initial congestion
        window of 1--2 packets.

      + Limited Transmit was used in over 20% of the servers.

      + While some servers do not use fast retransmit the
        overwhelming majority implement it.

      + Many web servers use the fast recovery algorithm (with a
        number using more advanced recovery such as NewReno
        [RFC3782] or SACK-based loss recovery techniques
        [RFC2018,RFC3517].

    (Note that [MAF05] updates some of the results of [PF01].  The
    newer results confirm the older results.)

References:

  [MAF05] Alberto Medina, Mark Allman, Sally Floyd.  Measuring the
    Evolution of Transport Protocols in the Internet.  ACM Computer
    Communication Review, 35(2), April 2005.

  [Pax97] Vern Paxson.  Automated Packet Trace Analysis of TCP
    Implementations.  ACM SIGCOMM, September 1997.

  [PF01] Jitu Padhye, Sally Floyd.  Identifying the TCP Behavior of
  Web Servers, SIGCOMM 2001, August 2001. 

  [WS95] Wright, G. and W. Stevens, "TCP/IP Illustrated, Volume 2: The
    Implementation", Addison-Wesley, 1995.

Attachment 1:

  Date:    Mon, 24 Sep 2007 19:55:20 PDT
  To:      mallman@icir.org
  From:    Stephen Hemminger <shemminger@linux-foundation.org>
  Subject: Re: rfc2581

  [...]

  Yes Linux implements RFC2581 and has not had any unstable or
  congestion problems caused by that. In recent years, there has
  been lots of refinements and alternatives added, but all the other
  algorithms are more complex attempts to ensure proper and stable
  response in "corner case" domains of large delay bandwidth
  products and/or small router queues.

  Linux also implements RFC2861 (congestion window validation) by
  default which makes it less aggressive than many other
  implementations. Because this caused some bursty applications to
  have poor performance it was made optional.

  The only real complaint against the principles of congestion
  control has come from the financial community. Slow start can
  cause connections to have latency, and when latency equates to
  real $$ during transactions, customers get very sensitive to the
  added delay.  For a discussion of this see the presentation from
  Credit Suisse at this 2007 Kernel
  Summit. http://lwn.net/Articles/248878/ For that reason, they are
  looking to alternatives to TCP/IP such as Infiniband.

  -- 
  Stephen Hemminger <shemminger@linux-foundation.org>

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From: Lars Eggert <lars.eggert@nokia.com>
Subject: Re: [tcpm] WGLC for UTO
Date: Wed, 31 Oct 2007 09:47:37 +0200
To: ext Joe Touch <touch@ISI.EDU>
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On 2007-10-30, at 18:07, ext Joe Touch wrote:
> Lars Eggert wrote:
>> The intent was to make UTO work with SYN cookie implementations that
>> can't encode whether a UTO option was present in the cookie, by
>> including the option also in the first non-SYN packets.
>
> That should be deemed non-compliant. The point of the SYN cookie is to
> reestablish the entirety of state that would have been established by
> the SYN; if the cookie can't do that, a cookie should not be used.

Well, if the SYN requests (many) options and the receiver cannot  
encode them all in the cookie, it can always reply with a SYN-ACK  
that only echoes those that it can and does encode. That's compliant,  
no?

> TCP has no provisions for post-SYN establishment of new connection  
> state
> AFAIK. Requiring this of this option would be new, and, IMO, a bit
> confusing for designers. What happens if the post-SYN packet lacks the
> option? do you declare state mis-synchronization and drop it?

No - the UTO option is purely advisory and not transmitted reliably.  
We had a long thread a while ago on the list that boiled down to that  
consensus. (I can dig it up if you like me to.) The gist was that it  
should be OK these days to send new options during a connection, even  
if the SYN exchange didn't negotiate their use.

>> So there is some use to doing this. We could state that
>> including UTo in the first non-SYN packets is only useful if the  
>> option
>> was not included in the SYN?
>
> I'd prefer that, adjusted to say "if the option was not included in  
> the
> SYN due to space limitations".

That can certainly be done.

>>> - sec 3 "A host that supports"
>>> What happens if there is no option space in the next outgoing  
>>> segment?
>>> What happens if there is no option space at all? Do you tell the  
>>> app? Do
>>> you do anything special? IMO, this deserves a separate section  
>>> rather
>>> than just a note at the end of 3.0
>>
>> I don't know what we can say about this - this is a general issue  
>> with
>> TCP. What should be done depends a lot on which other options are  
>> in use
>> for a particular connection. Maybe UTO is considered more  
>> important than
>> some options, but less important than others. Which options do you
>> include and which ones do you not include, generally? Hard to say.
>
> I'm not making an assertion as to what to do, only that the doc  
> ought to
> consider the issue. I'll suggest that it ought to send it within 2
> segments OR indicate to the application that it has failed; the latter
> requires a new API variable, though.

We can add that TCP should try for a few segments to include a TO  
option if it wants to send one, but I don't think notifying the  
application on failure is necessary, because UTO options are advisory  
and not transmitted reliably. Not being able to include it locally is  
equivalent to having the segment that it was included on being  
dropped in the network.

>>> - sec 3.1 "In general"
>>> This is contrary to the default values. CHANGEABLE defaults to  
>>> true, but
>>> ENABLED defaults to false. That means, IMO, that received UTO  
>>> options
>>> would be dropped, not acted upon, in the default case.
>>
>> When the draft was targeted as Experimental, having ENABLED  
>> default to
>> false was needed, so that hosts would need to actively enable the
>> mechanism to participate to experiment with the option.
>>
>> If we're going with PS (are we?), ENABLED should default to true,  
>> I guess.
>
> Sure - making the two consistent is the issue.

Will do, as soon as the chairs declare consensus on which way they  
see the WG consensus go.

> ...
>>> Very minor points, if an edit occurs:
>> ...
>>
>> Most of these are addressed in the next revision. But not these,  
>> so far:
>>
>>> - sec 3.1 "This means that"
>>> This paragraph is part of a discussion on checks and limits; this  
>>> should
>>> not be the first place you introduce the "max" concept.
>>
>> It's not - the formula further up introduces it.
>
> It might be useful to introduce the issue in prose first. The  
> reason for
> using max is the issue, and it's not clear by just stating the  
> formula.
>
>>> - hysteresis?
>>> The doc doesn't describe how quickly the UTO can be changed by the
>>> remote end or application. Does it matter, e.g., for security  
>>> reasons?
>>
>> I don't think there are security implications here. The only  
>> result of
>> changing the value very frequently is that all segments have the  
>> option
>> attached, so there is some mild increase in bandwidth use. But the  
>> app
>> can always just send more data anyway :-)
>
> Can someone on the path do something you don't want to your connection
> by tacking this option on?

So that's a different question than your original one, which was  
changing the value very frequently.

An attacker that adds or modifies the content of UTO options may be  
able to affect how long the ends keep the connection alive. Since UTO  
is advisory, an attacker can't really do anything to the connection  
that couldn't happen anyway. And the security properties of UTO are  
similar to other TCP options (window scaling, timestamps), in that  
none of them are authenticated or encrypted.

>>> - sec 3.2
>>> If you retransmit the option and attach it to a particular byte,  
>>> and the
>>> byte is ACKd, then why don't you KNOW the option has been received
>>> (i.e., second part of the handshake)? if the other end receives data
>>> over 1 window away from that byte, why doesn't it KNOW the source  
>>> end
>>> knows it has been recieved (i.e., third part of the handshake)?
>>
>> *If* you do this, yes, but there was consensus earlier that the WG
>> didn't want the UTO exchange to be reliable, which is why this
>> "attaching" the option to a byte is a MAY.
>
> Ah - it might be useful to explain that. I thought MAY implied that  
> the
> information about reliability was in doubt, rather than that the  
> use of
> the attachment was optional but would provide reliable UTO exchange  
> if used.

The MAY is about that even though the spec says that an UTO is  
transmitted unreliably, *if* an implementation really wanted to, it  
could try to implement some local mechanisms to make transmission of  
its UTOs to the peer more reliable. (But it can't enforce reliability  
in the other direction.)

>>> About retransmission, IMO, it SHOULD be done - or else what's the  
>>> point
>>> of this?
>>
>> Is this related to your previous comment? If so, there was  
>> consensus to
>> not make UTO reliable, i.e., retransmit lost options.
>
> I have no idea what the point of an unreliable option on a reliable
> transport protocol means. IMO, options should be retransmitted when
> their associated segments are. The control protocol ought to be at  
> least
> as robust as the data. Although I appreciate that UTO isn't as  
> critical
> as some other options or flags, introducing nondeterminism at this  
> level
> doesn't seem appropriate, IMO.

We had a long discussion on this a year or two ago on this list. The  
outcome was that because UTO is advisory, a simple, unreliable  
transmission is sufficient. We can certainly revisit this decision,  
but I'd personally would like to see strong WG consensus on changing  
this pretty central assumption behind UTO this late in the game.

> I.e., I think there are separate questions:
>
> 	1) is UTO critical enough to ensure correct exchange via
> 	an explicitly ack'd handshake
> 		no - I agree with the WG
>
> 	2) should options associated with a segment be retransmitted
> 	with that segment in general in TCP
> 		I think the answer to this should be YES, always,
> 		but I don't think it was asked.

Lars
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Lars Eggert wrote:
> On 2007-10-30, at 18:07, ext Joe Touch wrote:
>> Lars Eggert wrote:
>>> The intent was to make UTO work with SYN cookie implementations that
>>> can't encode whether a UTO option was present in the cookie, by
>>> including the option also in the first non-SYN packets.
>>
>> That should be deemed non-compliant. The point of the SYN cookie is to=

>> reestablish the entirety of state that would have been established by
>> the SYN; if the cookie can't do that, a cookie should not be used.
>=20
> Well, if the SYN requests (many) options and the receiver cannot encode=

> them all in the cookie, it can always reply with a SYN-ACK that only
> echoes those that it can and does encode. That's compliant, no?

It is, but only if current SYN cookie responses for stacks that don't
support UTO could be detected. I'm not sure that's possible.

>> TCP has no provisions for post-SYN establishment of new connection sta=
te
>> AFAIK. Requiring this of this option would be new, and, IMO, a bit
>> confusing for designers. What happens if the post-SYN packet lacks the=

>> option? do you declare state mis-synchronization and drop it?
>=20
> No - the UTO option is purely advisory and not transmitted reliably. We=

> had a long thread a while ago on the list that boiled down to that
> consensus. (I can dig it up if you like me to.) The gist was that it
> should be OK these days to send new options during a connection, even i=
f
> the SYN exchange didn't negotiate their use.

Then there's no point in having the SYN cookie encode anything.
Receivers that implement SYN cookies MAY drop the option; senders
sending UTO have no reason to resend, since it's 'advisory and not
transmitted reliably'.

I.e., you can't have it both ways. If this is truly unreliable, then
there is NO reason to ever retransmit it.

=2E..
> We can add that TCP should try for a few segments to include a TO optio=
n
> if it wants to send one, but I don't think notifying the application on=

> failure is necessary, because UTO options are advisory and not
> transmitted reliably. Not being able to include it locally is equivalen=
t
> to having the segment that it was included on being dropped in the netw=
ork.

Again, if that's the case, then try to send it ONCE. Never retry -
there's no point.

=2E..
>> Can someone on the path do something you don't want to your connection=

>> by tacking this option on?
>=20
> So that's a different question than your original one, which was
> changing the value very frequently.
>=20
> An attacker that adds or modifies the content of UTO options may be abl=
e
> to affect how long the ends keep the connection alive. Since UTO is
> advisory, an attacker can't really do anything to the connection that
> couldn't happen anyway.=20

I'm not sure that's true; even though it's advisory, it can cause a
change in behavior - if not, then there's no point in sending it. An
attacker could cause large UTOs for receivers that act on the option,
which could cause a DOS opportunity.

> And the security properties of UTO are similar
> to other TCP options (window scaling, timestamps), in that none of them=

> are authenticated or encrypted.

Yes, but the implications of changing these others are different - and
each should be explained in their respective security sections. This
security section should say what happens when an attacker changes THIS
option.

=2E..
>>>> About retransmission, IMO, it SHOULD be done - or else what's the po=
int
>>>> of this?
>>>
>>> Is this related to your previous comment? If so, there was consensus =
to
>>> not make UTO reliable, i.e., retransmit lost options.
>>
>> I have no idea what the point of an unreliable option on a reliable
>> transport protocol means. IMO, options should be retransmitted when
>> their associated segments are. The control protocol ought to be at lea=
st
>> as robust as the data. Although I appreciate that UTO isn't as critica=
l
>> as some other options or flags, introducing nondeterminism at this lev=
el
>> doesn't seem appropriate, IMO.
>=20
> We had a long discussion on this a year or two ago on this list. The
> outcome was that because UTO is advisory, a simple, unreliable
> transmission is sufficient. We can certainly revisit this decision, but=

> I'd personally would like to see strong WG consensus on changing this
> pretty central assumption behind UTO this late in the game.

Agreed.

Thanks,

Joe


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Subject: RE: [tcpm] WGLC for UTO
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-----Original Message-----
From: Lars Eggert [mailto:lars.eggert@nokia.com]
Sent: Wed 10/31/2007 12:47 AM
To: ext Joe Touch
Cc: tcpm@ietf.org
Subject: Re: [tcpm] WGLC for UTO
=20
On 2007-10-30, at 18:07, ext Joe Touch wrote:
> Lars Eggert wrote:
>> The intent was to make UTO work with SYN cookie implementations that
>> can't encode whether a UTO option was present in the cookie, by
>> including the option also in the first non-SYN packets.
>
> That should be deemed non-compliant. The point of the SYN cookie is to
> reestablish the entirety of state that would have been established by
> the SYN; if the cookie can't do that, a cookie should not be used.

Well, if the SYN requests (many) options and the receiver cannot =20
encode them all in the cookie, it can always reply with a SYN-ACK =20
that only echoes those that it can and does encode. That's compliant, =20
no?

----------- End of Original Message ---------------------

Compliant, yes, but not desirable. It would mean that any system
that had enabled SYN Cookie mode would stop accepting UTOs.
This is not a very intuitive model for application developers, and
would likely just add one more log to the "UTO is not reliable" fire
and discourage it's use.

What I think we want is for any system in the passive role that is
*capable* of doing UTO to refrain from doing any action that would
block enabling UTO on the first non-SYN received.

It is especially important that the passive side not be *obligated*
to remember that there was no UTO option in the SYN packet, as
in *expecting* it to reject later packets for having improper options.
That would essentially mean that the system would have to disable
UTO *permanently* as soon as it enabled SYN Cookies once (unless
it was actually going to track which connections were established
when and check that in real-time).

My understanding of including the UTO in both the SYN as well
as the first non-SYN was that network monitoring equipment (or
software) would expect any negotiated option to have been present
in the SYN and likely would not know that UTO was an exception.


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<P><FONT SIZE=3D2>-----Original Message-----<BR>
From: Lars Eggert [<A =
HREF=3D"mailto:lars.eggert@nokia.com">mailto:lars.eggert@nokia.com</A>]<B=
R>
Sent: Wed 10/31/2007 12:47 AM<BR>
To: ext Joe Touch<BR>
Cc: tcpm@ietf.org<BR>
Subject: Re: [tcpm] WGLC for UTO<BR>
<BR>
On 2007-10-30, at 18:07, ext Joe Touch wrote:<BR>
&gt; Lars Eggert wrote:<BR>
&gt;&gt; The intent was to make UTO work with SYN cookie implementations =
that<BR>
&gt;&gt; can't encode whether a UTO option was present in the cookie, =
by<BR>
&gt;&gt; including the option also in the first non-SYN packets.<BR>
&gt;<BR>
&gt; That should be deemed non-compliant. The point of the SYN cookie is =
to<BR>
&gt; reestablish the entirety of state that would have been established =
by<BR>
&gt; the SYN; if the cookie can't do that, a cookie should not be =
used.<BR>
<BR>
Well, if the SYN requests (many) options and the receiver =
cannot&nbsp;<BR>
encode them all in the cookie, it can always reply with a =
SYN-ACK&nbsp;<BR>
that only echoes those that it can and does encode. That's =
compliant,&nbsp;<BR>
no?<BR>
<BR>
----------- End of Original Message ---------------------<BR>
<BR>
Compliant, yes, but not desirable. It would mean that any system<BR>
that had enabled SYN Cookie mode would stop accepting UTOs.<BR>
This is not a very intuitive model for application developers, and<BR>
would likely just add one more log to the &quot;UTO is not =
reliable&quot; fire<BR>
and discourage it's use.<BR>
<BR>
What I think we want is for any system in the passive role that is<BR>
*capable* of doing UTO to refrain from doing any action that would<BR>
block enabling UTO on the first non-SYN received.<BR>
<BR>
It is especially important that the passive side not be *obligated*<BR>
to remember that there was no UTO option in the SYN packet, as<BR>
in *expecting* it to reject later packets for having improper =
options.<BR>
That would essentially mean that the system would have to disable<BR>
UTO *permanently* as soon as it enabled SYN Cookies once (unless<BR>
it was actually going to track which connections were established<BR>
when and check that in real-time).<BR>
<BR>
My understanding of including the UTO in both the SYN as well<BR>
as the first non-SYN was that network monitoring equipment (or<BR>
software) would expect any negotiated option to have been present<BR>
in the SYN and likely would not know that UTO was an exception.<BR>
<BR>
</FONT>
</P>

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From: Lars Eggert <lars.eggert@nokia.com>
Subject: Re: [tcpm] WGLC for UTO
Date: Wed, 31 Oct 2007 18:30:47 +0200
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On 2007-10-31, at 17:54, ext Joe Touch wrote:
> Lars Eggert wrote:
>> On 2007-10-30, at 18:07, ext Joe Touch wrote:
>>> Lars Eggert wrote:
>>>> The intent was to make UTO work with SYN cookie implementations  
>>>> that
>>>> can't encode whether a UTO option was present in the cookie, by
>>>> including the option also in the first non-SYN packets.
>>>
>>> That should be deemed non-compliant. The point of the SYN cookie  
>>> is to
>>> reestablish the entirety of state that would have been  
>>> established by
>>> the SYN; if the cookie can't do that, a cookie should not be used.
>>
>> Well, if the SYN requests (many) options and the receiver cannot  
>> encode
>> them all in the cookie, it can always reply with a SYN-ACK that only
>> echoes those that it can and does encode. That's compliant, no?
>
> It is, but only if current SYN cookie responses for stacks that don't
> support UTO could be detected. I'm not sure that's possible.

Can you rephrase? I don't follow. (Maybe it's too late in the day here.)

>>> TCP has no provisions for post-SYN establishment of new  
>>> connection state
>>> AFAIK. Requiring this of this option would be new, and, IMO, a bit
>>> confusing for designers. What happens if the post-SYN packet  
>>> lacks the
>>> option? do you declare state mis-synchronization and drop it?
>>
>> No - the UTO option is purely advisory and not transmitted  
>> reliably. We
>> had a long thread a while ago on the list that boiled down to that
>> consensus. (I can dig it up if you like me to.) The gist was that it
>> should be OK these days to send new options during a connection,  
>> even if
>> the SYN exchange didn't negotiate their use.
>
> Then there's no point in having the SYN cookie encode anything.

If the SYN cookie could always encode the UTO value, we didn't need  
to also include a UTO with the first non-SYN packet. But I don't know  
of any SYN cookie encoding that can. (And SYN cookies aren't  
standardized anyway.)

All this is a very narrow corner case anyway. The main motivation for  
sending UTO twice (in the SYN and first non-SYN segment) was to make  
sure it takes effect as early as possible in the connection. If this  
is a showstopper, we could remove the recommendation to send it in  
the SYNs and only recommend to send it in the first non-SYN.

> Receivers that implement SYN cookies MAY drop the option; senders
> sending UTO have no reason to resend, since it's 'advisory and not
> transmitted reliably'.
>
> I.e., you can't have it both ways. If this is truly unreliable, then
> there is NO reason to ever retransmit it.

The reason we special-cased this is that we know of a case (SYN  
cookies) where a UTO on the SYN will be "dropped" (because it can't  
be encoded). As I said above, we can remove this special case, if  
people feel this is cleaner.

>> We can add that TCP should try for a few segments to include a TO  
>> option
>> if it wants to send one, but I don't think notifying the  
>> application on
>> failure is necessary, because UTO options are advisory and not
>> transmitted reliably. Not being able to include it locally is  
>> equivalent
>> to having the segment that it was included on being dropped in the  
>> network.
>
> Again, if that's the case, then try to send it ONCE. Never retry -
> there's no point.

It was YOUR suggestion in an earlier email to try and include a UTO  
in one of a small sequence of a few consecutive segments if option  
space limitation prevented sending it on the next outgoing segment.  
The draft doesn't have this.

>>> Can someone on the path do something you don't want to your  
>>> connection
>>> by tacking this option on?
>>
>> So that's a different question than your original one, which was
>> changing the value very frequently.
>>
>> An attacker that adds or modifies the content of UTO options may  
>> be able
>> to affect how long the ends keep the connection alive. Since UTO is
>> advisory, an attacker can't really do anything to the connection that
>> couldn't happen anyway.
>
> I'm not sure that's true; even though it's advisory, it can cause a
> change in behavior - if not, then there's no point in sending it. An
> attacker could cause large UTOs for receivers that act on the option,
> which could cause a DOS opportunity.

Sure, but nothing in TCP is resistant against on-path attackers. We  
can certainly mention that there is a danger, but an on-path attacker  
can do all sorts of other things to a connection.

>>>>> About retransmission, IMO, it SHOULD be done - or else what's  
>>>>> the point
>>>>> of this?
>>>>
>>>> Is this related to your previous comment? If so, there was  
>>>> consensus to
>>>> not make UTO reliable, i.e., retransmit lost options.
>>>
>>> I have no idea what the point of an unreliable option on a reliable
>>> transport protocol means. IMO, options should be retransmitted when
>>> their associated segments are. The control protocol ought to be  
>>> at least
>>> as robust as the data. Although I appreciate that UTO isn't as  
>>> critical
>>> as some other options or flags, introducing nondeterminism at  
>>> this level
>>> doesn't seem appropriate, IMO.
>>
>> We had a long discussion on this a year or two ago on this list. The
>> outcome was that because UTO is advisory, a simple, unreliable
>> transmission is sufficient. We can certainly revisit this  
>> decision, but
>> I'd personally would like to see strong WG consensus on changing this
>> pretty central assumption behind UTO this late in the game.
>
> Agreed.

Noted.

Thanks,
Lars
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